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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Hi,

I am having an issue with Asterisk 11.18.0 where after a certain amount of queue calls (I would say about 40), the system stops responding to inbound calls. It doesn't actually ever crash but I can see many locks (Core show locks) in the system. Some are related to mysql. I have an external system that is querying the database and it never stops functioning therefore I don't believe it is a mysql issue.

The only errors I can see in the log are:
Code: exceptionally long voice queue length (this error starts showing up under heavy load).


htop and top show only 20% cpu usage during the occurrence of the issue and RAM does not seem to be an issue either.

Running Centos 6.8 32bit. 8GB RAM (PAE). Quad Core processor with HT x5550 2.67Ghz.

Any suggestions? What else can I use to troubleshoot?

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781
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Taking one recording for Transfer call.

July 8, 2015, 1:11 am
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Hi guys,
I am using Elastix 2.4, when a call is transfer to any other extension the recording splits in to two sections. From the source extension to called extension and then from called extension to transferred extension. Can we make it a one recording, like if a call is transferred two or three extensions it show only one recording in panel as well as in server path

Cheers,
MB

Statistics : Posted by muhammadbilal • on Wed Jul 08, 2015 1:11 am • Replies 1 • Views 162
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res_ari_mailboxes.so: undefined symbol

July 15, 2015, 2:41 pm
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jcolp wrote:The error you mention is actually a warning, it occurs because app_voicemail and external mailbox support can't co-exist. As res_ari_mailboxes depends on external mailbox support and it is not present, it warns that it can't load.

What exact problem are you having that needs solving?


The problem is that the phones are not reachable and immediately go busy when trying to dial out. After some fiddling, I either broke dahdi or it's somehow not loading properly:

Code: [Jul 17 14:16:32] WARNING[7259][C-00000000] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)


I think it would be better if I posted a new message for this. Thanks so much.

Statistics : Posted by gossamer • on Wed Jul 15, 2015 2:41 pm • Replies 2 • Views 177
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res_ari_mailboxes.so: undefined symbol

July 15, 2015, 2:41 pm
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The error you mention is actually a warning, it occurs because app_voicemail and external mailbox support can't co-exist. As res_ari_mailboxes depends on external mailbox support and it is not present, it warns that it can't load.

What exact problem are you having that needs solving?

Statistics : Posted by gossamer • on Wed Jul 15, 2015 2:41 pm • Replies 2 • Views 177
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res_ari_mailboxes.so: undefined symbol

July 15, 2015, 2:41 pm
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Hi, I'm using the asterisk-13.1.1 RPM included with fedora22 with dahdi downloaded and compiled from git and receiving the following error:

Code: [Jul 15 11:25:51] WARNING[31751] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 15 11:25:51] WARNING[31751] loader.c: Module 'res_ari_mailboxes.so' could not be loaded.


This is upgraded from fedora21 where it all worked properly. Is this a fedora RPM problem or an asterisk problem? I'd rather not have to compile that from source too.

I also have a number of other errors/warnings, and I don't know if they're related, but I'm really stuck. These are from the messsages file when starting asterisk. Where do I go about troubleshooting these problems?

The actual problem I'm having is that asterisk answers the calls but cannot communicate with any IP phones.

Code: [Jul 15 11:25:50] Asterisk 13.1.1 built by mockbuild @ buildhw-06.phx2.fedoraproject.org on a x86_64 running Linux on 2015-01-30 20:30:35 UTC
[Jul 15 11:25:51] NOTICE[31751] cdr.c: CDR simple logging enabled.
[Jul 15 11:25:51] NOTICE[31751] loader.c: 220 modules will be loaded.
[Jul 15 11:25:51] WARNING[31751] res_phoneprov.c: Unable to find a valid server address or name.
[Jul 15 11:25:51] WARNING[31751] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse20060419.pub: Permission denied
[Jul 15 11:25:51] WARNING[31751] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse01.pub: Permission denied
[Jul 15 11:25:51] ERROR[31751] ari/config.c: No configured users for ARI
[Jul 15 11:25:51] WARNING[31751] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 15 11:25:51] WARNING[31751] loader.c: Module 'res_ari_mailboxes.so' could not be loaded.
[Jul 15 11:25:51] WARNING[31751] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Jul 15 11:25:52] NOTICE[31751] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Jul 15 11:25:52] NOTICE[31751] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 15 11:25:52] WARNING[31751] pbx.c: Unable to register extension '_X' priority 1 in 'mainmenu-night', already in use
[Jul 15 11:25:52] WARNING[31751] pbx_config.c: Unable to register extension at line 220 of extensions.conf
[Jul 15 11:25:52] WARNING[31751] pbx.c: Unable to register extension 's' priority 2 in 'incoming-sales', already in use
[Jul 15 11:25:52] WARNING[31751] pbx_config.c: Unable to register extension at line 292 of extensions.conf
[Jul 15 11:25:52] ERROR[31751] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[Jul 15 11:25:52] WARNING[31751] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages


Statistics : Posted by gossamer • on Wed Jul 15, 2015 2:41 pm • Replies 2 • Views 177
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Calls going to from-sip-external by default from Java code

July 17, 2015, 7:15 am
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Hi David,

The line in the code:

originateAction.setChannel("SIP/10001@xxx.xxx.xxx.xxx");

The ip address specified is the IP address of the Asterisk Server. Earlier i had tried using no IP Address at all while setting the channel but the code compiled with an error.

And while using the softphone to add and register a new user with id 10001, i use the same string format as "10001@xxx.xxx.xxx.xxx".

So, please let me know if i am wrong in my observation or something else is wrong.

Thanks.

Statistics : Posted by manas539 • on Fri Jul 17, 2015 7:15 am • Replies 1 • Views 68
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Calls going to from-sip-external by default from Java code

July 17, 2015, 7:15 am
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Looks like you have specified your own address in the destination channel address specification and the call is looped back.

Statistics : Posted by manas539 • on Fri Jul 17, 2015 7:15 am • Replies 1 • Views 68
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Calls going to from-sip-external by default from Java code

July 17, 2015, 7:15 am
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Hi all,

I am a newbie to Asterisk. I am using the Asterisk-JAVA API to call Asterisk server using an extension and a setup account. Using the code and the configuration i am able to call the desired extension successfully from a softphone.But when i try to do the same using the JAVA code then the call gets processed by the "from-external-sip" extension and not by the extension i have setup in the code.Below is the code :
Code: import java.io.IOException;

import org.asteriskjava.*;
import org.asteriskjava.manager.*;
import org.asteriskjava.manager.action.OriginateAction;
import org.asteriskjava.manager.response.ManagerResponse;
import org.asteriskjava.manager.AuthenticationFailedException;
import org.asteriskjava.manager.ManagerConnection;
import org.asteriskjava.manager.ManagerConnectionFactory;
import org.asteriskjava.manager.TimeoutException;


public class TryConnect  {
    private ManagerConnection managerConnection;

    public TryConnect() throws IOException
    {
       ManagerConnectionFactory factory = new ManagerConnectionFactory("xxx.xxx.xxx.xxx", "xxx", "xxxx");
       this.managerConnection = factory.createManagerConnection();
              
    }

    public void run() throws IOException, AuthenticationFailedException,TimeoutException
    {
        OriginateAction originateAction;
        ManagerResponse originateResponse;

        originateAction = new OriginateAction();
         
        originateAction.setChannel("SIP/10001@xxx.xxx.xxx.xxx");
       
        originateAction.setContext("from-internal");
        originateAction.setExten("1000");
        originateAction.setCallerId("1000");
       
       
        originateAction.setPriority(new Integer(1));
        originateAction.setTimeout(new Integer(5000));
       

        // connect to Asterisk and log in
          managerConnection.login();
       

        // send the originate action and wait for a maximum of 30 seconds for Asterisk
        // to send a reply
        originateResponse = managerConnection.sendAction(originateAction, 5000);

        // print out whether the originate succeeded or not
        System.out.println(originateResponse.getResponse());

        // and finally log off and disconnect
        managerConnection.logoff();
    }

    public static void main(String[] args) throws Exception
    {
       TryConnect helloManager;

        helloManager = new TryConnect();
        helloManager.run();
    }
}



Below is the output in the CLI on making the call from the JAVA code :-
Code: -- Executing [10001@from-sip-external:1] NoOp("SIP/192.168.220.10-00000007", "Received incoming SIP connection from unknown peer to 10001") in new stack
    -- Executing [10001@from-sip-external:2] Set("SIP/192.168.220.10-00000007", "DID=10001") in new stack
    -- Executing [10001@from-sip-external:3] Goto("SIP/192.168.220.10-00000007", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.220.10-00000007", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/192.168.220.10-00000007", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2015-07-17 00:04:16.259 CEST.
    -- Executing [s@from-sip-external:6] Log("SIP/192.168.220.10-00000007", "WARNING,"Rejecting unknown SIP connection from 192.168.220.10"") in new stack
    -- Executing [s@from-sip-external:7] Answer("SIP/192.168.220.10-00000007", "") in new stack
    -- Executing [1000@from-internal:1] Dial("SIP/192.168.220.10-00000006", "SIP/1000") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1000@from-internal:2] Answer("SIP/192.168.220.10-00000006", "") in new stack
    -- Executing [1000@from-internal:3] Playback("SIP/192.168.220.10-00000006", "pls-try-call-later") in new stack
    -- Executing [s@from-sip-external:8] Wait("SIP/192.168.220.10-00000007", "2") in new stack
    -- <SIP/192.168.220.10-00000006> Playing 'pls-try-call-later.ulaw' (language 'en')
  == Manager 'admin' logged off from 192.168.220.20
    -- Executing [s@from-sip-external:9] Playback("SIP/192.168.220.10-00000007", "ss-noservice") in new stack
    -- <SIP/192.168.220.10-00000007> Playing 'ss-noservice.ulaw' (language 'en')
    -- Executing [1000@from-internal:4] Hangup("SIP/192.168.220.10-00000006", "") in new stack
  == Spawn extension (from-internal, 1000, 4) exited non-zero on 'SIP/192.168.220.10-00000006'
    -- Executing [h@from-internal:1] Hangup("SIP/192.168.220.10-00000006", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/192.168.220.10-00000006'
  == Spawn extension (from-sip-external, s, 9) exited non-zero on 'SIP/192.168.220.10-00000007'
    -- Executing [h@from-sip-external:1] NoOp("SIP/192.168.220.10-00000007", "Received incoming SIP connection from unknown peer to h") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/192.168.220.10-00000007", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/192.168.220.10-00000007", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.220.10-00000007", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/192.168.220.10-00000007", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2015-07-17 00:04:18.584 CEST.
    -- Executing [s@from-sip-external:6] Log("SIP/192.168.220.10-00000007", "WARNING,"Rejecting unknown SIP connection from 192.168.220.10"") in new stack
    -- Executing [s@from-sip-external:7] Answer("SIP/192.168.220.10-00000007", "") in new stack
  == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/192.168.220.10-00000007'



Please help me out why the call is getting directed to the "from-external-sip" extension and not the "from-internal" extension as i have set up in the code.

Thanks.

Statistics : Posted by manas539 • on Fri Jul 17, 2015 7:15 am • Replies 1 • Views 68
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Unable to create channel of type 'DAHDI' (Cause 0 - Unknown)

July 17, 2015, 3:15 pm
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Hi,

I've recently upgraded asterisk to 13.1.1 on fedora22 from fedora20 and it's now not working. I've downloaded and compiled dahdi-linux from git, and the kernel modules are installed, but apparently asterisk now doesn't see them. It appears to detect my IP450 phone, but immediately goes busy when trying to dial out. Asterisk also doesn't answer when called.

The only thing I can think to do is paste the logs and my relevant config files in hopes that someone can help get my phone system running again.

Code: [Jul 17 16:36:41] Asterisk 13.1.1 built by mockbuild @ buildhw-06.phx2.fedoraproject.org on a x86_64 running Linux on 2015-01-30 20:30:35 UTC
[Jul 17 16:36:41] NOTICE[22326] cdr.c: CDR simple logging enabled.
[Jul 17 16:36:41] NOTICE[22326] loader.c: 221 modules will be loaded.
[Jul 17 16:36:41] WARNING[22326] res_phoneprov.c: Unable to find a valid server address or name.
[Jul 17 16:36:41] WARNING[22326] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse20060419.pub: Permission denied
[Jul 17 16:36:41] WARNING[22326] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse01.pub: Permission denied
[Jul 17 16:36:41] ERROR[22326] ari/config.c: No configured users for ARI
[Jul 17 16:36:41] WARNING[22326] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 17 16:36:41] WARNING[22326] loader.c: Module 'res_ari_mailboxes.so' could not be loaded.
[Jul 17 16:36:42] WARNING[22326] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Unable to specify channel 1: No such device or address
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to open channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to register channel '1'
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Channel '1' failure ignored: ignore_failed_channels.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Unable to specify channel 2: No such device or address
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to open channel 2: No such device or address
here = 0, tmp->channel = 2, channel = 2
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to register channel '2'
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Channel '2' failure ignored: ignore_failed_channels.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Unable to specify channel 3: No such device or address
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to open channel 3: No such device or address
here = 0, tmp->channel = 3, channel = 3
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to register channel '3'
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Channel '3' failure ignored: ignore_failed_channels.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Unable to specify channel 4: No such device or address
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to open channel 4: No such device or address
here = 0, tmp->channel = 4, channel = 4
[Jul 17 16:36:42] ERROR[22326] chan_dahdi.c: Unable to register channel '4'
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Channel '4' failure ignored: ignore_failed_channels.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Jul 17 16:36:42] WARNING[22326] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Jul 17 16:36:42] NOTICE[22358] chan_sip.c: Peer '7003' is now Reachable. (15ms / 2000ms)
[Jul 17 16:36:42] NOTICE[22326] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Jul 17 16:36:42] NOTICE[22326] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 17 16:36:42] WARNING[22326] pbx.c: Unable to register extension '_X' priority 1 in 'mainmenu-night', already in use
[Jul 17 16:36:42] WARNING[22326] pbx_config.c: Unable to register extension at line 220 of extensions.conf
[Jul 17 16:36:42] WARNING[22326] pbx.c: Unable to register extension 's' priority 2 in 'incoming-sales', already in use
[Jul 17 16:36:42] WARNING[22326] pbx_config.c: Unable to register extension at line 292 of extensions.conf
[Jul 17 16:36:42] ERROR[22326] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[Jul 17 16:36:42] WARNING[22326] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages
[Jul 17 16:36:50] ERROR[22394][C-00000000] utils.c: write() returned error: Broken pipe
[Jul 17 16:36:50] ERROR[22394][C-00000000] utils.c: write() returned error: Broken pipe
[Jul 17 16:36:50] WARNING[22394][C-00000000] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)


Code: # lsmod|egrep 'dahdi|wctdm'
wctdm                  49152  0
dahdi_echocan_mg2      16384  0
dahdi                 229376  2 wctdm,dahdi_echocan_mg2
crc_ccitt              16384  1 dahdi


Code: orion*CLI> module load chan_dahdi
Unable to load module chan_dahdi
Command 'module load chan_dahdi' failed.
[Jul 17 17:08:10] WARNING[25776]: loader.c:1028 load_resource: Module 'chan_dahdi' already exists.


Code: orion*CLI> dahdi show status
Description                              Alarms  IRQ    bpviol CRC    Fra Codi Options  LBO
Wildcard TDM400P REV I Board 5           UNCONFI 0      0      0      CAS Unk           0 db (CSU)/0-133 feet (DSX-1)


Code: [root@orion asterisk]# egrep -v '^#|^$|\;' sip.conf
[general]
port            = 5060
bindaddr        = 192.168.1.1
context         = incoming
limitonpeers    = yes
deny            = 0.0.0.0/0.0.0.0
permit          = 64.1.16.0/255.255.255.224
disallow        = all
allow           = ulaw
[7003]
type            = friend
accountcode     = dave
host            = dynamic
defaultuser     = Dave
secret          = Dave-7003
mailbox         = 7003@local
dtmfmode        = rfc2833
callerid        = Dave <7003>
context         = trusted
qualify         = yes
canreinvite     = no
call-limit      = 5


Code: [root@orion asterisk]# egrep -v '^#|^$|\;' chan_dahdi.conf
[channels]
language                        = en
context                         = incoming
accountcode                     = pstn-incoming
signalling                      = fxs_ks
callwaiting                     = no
immediate                       = no
amaflags                        = default
cancallforward                  = no
busydetect                      = no
callprogress                    = no
musiconhold                     = default
threewaycalling                 = no
callreturn                      = no
transfer                        = no
usedistinctiveringdetection     = no
usecallerid                     = yes
hidecallerid                    = no
callwaitingcallerid             = no
echocancel                      = yes
echotraining                    = no
group                           = 1
pickupgroup                     = 1
channel                         => 1
channel                         => 2
group                           =  2
context                         = incoming
channel                         => 3
group                           =  3
context                         = incoming
channel                         => 4


Code: [root@orion dahdi]# egrep -v '^#|^$|\;' system.conf
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
loadzone        = us
defaultzone     = us


Code: [root@orion asterisk]# rpm -qva|egrep 'asterisk|dahdi'
dahdi-tools-devel-2.10.0-2.fc22.x86_64
dahdi-tools-2.10.0-2.fc22.x86_64
asterisk-sounds-core-en-1.4.26-1.fc22.noarch
asterisk-voicemail-plain-13.1.1-1.fc22.x86_64
asterisk-dahdi-13.1.1-1.fc22.x86_64
asterisk-sounds-core-en-gsm-1.4.26-1.fc22.noarch
dahdi-tools-libs-2.10.0-2.fc22.x86_64
asterisk-gui-2.0-10.20120518svn5220.fc21.noarch
asterisk-13.1.1-1.fc22.x86_64
asterisk-voicemail-13.1.1-1.fc22.x86_64
asterisk-sip-13.1.1-1.fc22.x86_64


Thanks for any ideas. Please let me know what information I can provide to troubleshoot this.

Statistics : Posted by gossamer • on Fri Jul 17, 2015 3:15 pm • Replies 0 • Views 45
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INAP configuration with Digium TE420 card on Asterisk

July 17, 2015, 12:28 pm
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I suspect you have to purchase software that supports it.

For the hardware question, you should ask Digium the company, not the Asterisk opens source community.

However, if it is built on SS7, I imagine that it will differ at a level where the hardware is not involved.

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 1 • Views 72
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INAP configuration with Digium TE420 card on Asterisk

July 17, 2015, 12:28 pm
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I am currently working on Missed Call project in India. I have setup the whole project on Asterisk (using CentOS 6.6 as a base OS).We have deployed this project with different telecom provider on SIP, SS7 etc.

One of the telecom provider is providing us the connectivity on SS7-TDM using INAP protocol.They have provided us the GT and SPC . Now the challenge is we don't have any library which supports INAP. I have a few doubts which is listed below:
1) Does Digium card(TE420) supports INAP
2) Do We have to purchase some other hardware device to handle this INAP protocol.

Currently, we are getting following logs for the same configuration :

[1] Len = 196 [ 92 fe 3f c3 c0 0b cc 58 09 81 03 0e 19 0b 12 0c 00 12 04 19 87 63 39 99 81 0b 12 0c 00 12 04 19 89 19 30 00 01 9e 62 81
9b 48 04 5e c8 97 e2 6b 1e 28 1c 06 07 00 11 86 05 01 01 01 a0 11 60 0f 80 02 07 80 a1 09 06 07 04 00 01 01 01 00 00 6c 73 a1 71 02 01
00 02 01 00 30 69 80 02 03 78 82 06 01 10 bb 47 99 f2 83 07 03 11 99 11 00 35 84 85 01 0a 89 01 01 8a 08 04 93 19 89 19 30 00 60 af 30
30 0a 02 01 15 a1 05 04 03 00 00 08 30 0f 02 01 17 a1 0a 30 08 04 04 98 91 00 ee 05 00 30 08 02 01 31 a1 03 83 01 11 30 07 02 01 3b a1
02 30 00 97 02 91 81 9a 02 60 01 bb 05 80 03 80 90 a3 9c 01 03 ]
[1] FSN: 126 FIB 1
[1] BSN: 18 BIB 1
[1] <[0] MSU
[1] [ 92 fe 3f ]
[1] Network Indicator: 3 Priority: 0 User Part: SCCP (3)
[1] [ c3 ]
[1] OPC 9008 DPC 3008 SLS 5
[1] [ c0 0b cc 58 ]
[1]
[1] Unable to process message destined for userpart 3; dropping message


I hope, I have described you in details

Statistics : Posted by vivek_raj • on Fri Jul 17, 2015 12:28 pm • Replies 1 • Views 72
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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At the moment, the forum server's encryption is unsafe as far as Debian Iceweasel is concerned, which means that it is not easy for me to access it from home, where I have some time. Consequently you are going to have to do more of the leg work and actually work out the cycle of locks, or the locks that seem to be being held for too long, and the threads that seem to be running when you would expect them to be waiting.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Oh, and I removed CDR logging via odbc/mysql. It now logs directly to filesystem with csv files.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Hi, we had the same issue occur almost immediately. Have not had much time to come back here because we had chaos with the system going down during busy hours (100 concurrent calls). We had to move them to their old Asterisk server which does not suffer from this issue.

Things we have tried:

Installed Asterisk 11 on brand new hardware (HP Server, Dual Xeon [32 cores total], 96GB RAM, RAID 10 10k Hard Drives). I know, overkill. This was all we had available at the time. Same issue occurs with this new server. I can successfully reproduce the issue with sipp on a simple dialplan:

Code: [from-sip-test]
exten => 696969,1,Answer
exten => 696969,n,Musiconhold(default)
exten => 696969,4,Hangup


After about 30 calls, (10 calls per second), sipp starts receivinv transmit errors (no response from host).

command:
Quote:netstat -anNp | grep 5060


Shows thousands of packets queued.

Asterisk only shows about 10 calls.
See this in the error log:

Code: res_musiconhold.c: Failed to write frame to 'SIP/sip-test-00000024': Resource temporarily unavailable


Htop shows over 80GB of RAM free, CPU spikes to almost 95% during this time (all 32 cores).

Backtrace:
http://pastebin.com/qY6X5mqm

Core show locks:
http://pastebin.com/cUdFm5mB

I tried different Asterisk versions to 11.12, 11.16, same results. This was running on 11.18. Old server which does not suffer from this issue (as far as I can tell) is running 11.12 x64 on debian. This test was done on Ubuntu 14, 64 bit.

Not sure what else to try.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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I have now moved the mysql database and put it on it's own dedicated server. Let's see how it functions today.

Thanks

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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I agree, it looks as though something is up with the underlying database connection.

This is personally why I don't like tightly coupling Asterisk to databases. It's handy at times, but if your database acts up then Asterisk reflects that.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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You are looking for circles of two or more locks where threads are holding one and waiting on the other, or, in this case, signs that a lock is being held indefinitely for other reasons, namely the odbc lock at 0xab3e488 in this case.

It looks to me as though your ODBC connection for call logging has stalled and everything is waiting for it.

That's likely to be a third party problem and is one of the reasons why I would want to keep call logging as simple as possible.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Asterisk stops processing calls under heavy load

July 2, 2015, 8:56 pm
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Hi,

I am having an issue with Asterisk 11.18.0 where after a certain amount of queue calls (I would say about 40), the system stops responding to inbound calls. It doesn't actually ever crash but I can see many locks (Core show locks) in the system. Some are related to mysql. I have an external system that is querying the database and it never stops functioning therefore I don't believe it is a mysql issue.

The only errors I can see in the log are:
Code: exceptionally long voice queue length (this error starts showing up under heavy load).


htop and top show only 20% cpu usage during the occurrence of the issue and RAM does not seem to be an issue either.

Running Centos 6.8 32bit. 8GB RAM (PAE). Quad Core processor with HT x5550 2.67Ghz.

Any suggestions? What else can I use to troubleshoot?

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 14 • Views 957
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