[asterisk-users] Member stay busy after hangup a call in queue
Hello everyone. I am facing a problem with Asterisk version 1.6.2.24. What happens is that when you receive a call (A) in the queue and a member (B) answers the call normally. At the time that A or B off the member B of the queue continues as busy. The problem started occurring when my primary server had hardware failure and I had to migrate to a secondary server. The primary server was: Linux My-primary-server 2.6.26-2-686 # 1 SMP Thu Aug 19 03:44:10 UTC 2010 i686 GNU / Linux The secondary server is: Linux My-secondary-server 2.6.32-5-amd64 # 1 SMP Sun Sep 23 10:07:46 UTC 2012 x86_64 GNU / Linux On my primary server the problem did not occur and I used the version 1.6.2.21. By migrating to the secondary server I kept version 1.6.2.21 but the problem started to occur. I have reviewed the settings and found nothing wrong. So I decided to patch version 1.6.2.24 and the problem persisted. I removed the asterisk server and recompiled version 1.6.2.24 but the problem continued. And in the core show channels shows no channel which has problem. I know it is no longer provided support to version 1.6.2, but I need to make sure it's a bug in this version before migrating to version 1.8, it is a critical system. Any additional information that is required of the system I will provide. Any help is very welcome. Best regards, -- Rodrigo Lang http://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with macros in AEL
Hello list.
I am facing a small problem when I try to run a macro that is in AEL
through extensions.conf. I'm by applying Macro () invoking the macro, but
it always generates this message:
[09.20.2012 10:43:23] WARNING [28923] app_macro.c: No such context
'macro-dialout-trunk-building-custom-hook' for macro
'dialout-trunk-building-custom-hook'
And the macro is created like this:
macro-dialout-trunk-building {hook
Noop (--- Hi)
return;
}
I've tried to modify the macro name to macro-dialout-trunk-building-hook
me but the asterisk generates the following error:
[09.20.2012 11:19:10] WARNING [2721] app_macro.c: Context
'macro-dialout-trunk-building-hook' for macro 'dialout-trunk-building-hook'
lacks 's' extension, priority 1
I really do not know what else to do ... I'm even thinking of creating this
macro in conf, not only did it because she still has more than 500 lines.
I use FreePBX and Asterisk 1.8
Someone has a similar problem? Can anyone give me a help?
Thanks in advance!
--
Rodrigo Lang
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Problem with callfile and CDR
Hey, with the SIP works fine. Good tip.
But is this a bug with Local?
Thanks!
Rodrigo Lang.
2012/8/1 Danny Nicholas da...@debsinc.com
Just a WAG, but could the “local” channel be causing some kind of
problem? Perhaps if you changed local to SIP or DAHDI?
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, August 01, 2012 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Problem with callfile and CDR
** **
Good afternoon list.
** **
I am experiencing a problem with the CDR and callfiles. What is happening
is this: When generating a call with a callfile, everything works
perfectly, but the CDR is recorded in the table when they answer the call
destination. The field disposition is being recorded correctly, but the
duration field is marked with the ring time and billsec is marked with 0.
This just happens to connections through callfiles. Yes, the call is
working usually remains. I did several tests with durations from seconds to
20 minutes.
** **
I tested in two servers. With an Asterisk 1.4.44 (debian 5 64bits), and
another with Asterisk 1.8.14.0 (debian 6 64bits). In both occurs the same
problem. AND I tried using ForkCDR and ResetCDR and both did not help.
** **
I'm doing something wrong? Has anyone experienced something similar? Any
tips?
** **
** **
*The callfile:*
Channel: local/21411615@test_outgoing
CallerID: ELCO Test 123456789
MaxRetries: 1
RetryTime: 30
WaitTime: 25
Context: test_ivr
Extension: 21411615
Priority: 1
AlwaysDelete: Yes
Archive: Yes
** **
** **
*The extensions.conf*
** **
[test_outgoingsaida]
exten = _X.,1,Dial(khomp/gpstn/${EXTEN},120,Ttr)
exten = _X.,2,Hangup()
[test_ivr]
exten = _X.,1,Answer()
exten = _X.,n,Wait(20)
exten = _X.,n,Hangup()
** **
** **
*Example, console:*
** **
*Log first channel:*
[2012-08-01 14:29:44] DEBUG[6679] pbx.c: Launching 'Dial'
[2012-08-01 14:29:44] VERBOSE[6679] pbx.c: -- Executing
[21411615@test_outgoing:1] Dial(Local/21411615@test_outgoing-cb92;2,
khomp/gpstn/21411615,120,Ttr) in new stack
[2012-08-01 14:29:44] DEBUG[6679] devicestate.c: device 'Khomp/B1C0' state
'1'
[2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Can't find native
functions for channel 'Khomp/B1C0-0.0'
[2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Seeded SDP of
'Khomp/B1C0-0.0' with that of 'Local/21411615@test_outgoing-cb92;2'
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDTIME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
ANSWEREDTIME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDPEERNAME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDPEERNUMBER.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALSTATUS.
[2012-08-01 14:29:44] VERBOSE[6679] app_dial.c: -- Called
khomp/gpstn/21411615
[2012-08-01 14:29:52] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0 is
ringing
[2012-08-01 14:29:53] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0 is
making progress passing it to Local/21411615@test_outgoing-cb92;2
[2012-08-01 14:30:02] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0
answered Local/21411615@test_outgoing-cb92;2
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
write format slin
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
read format slin
[2012-08-01 14:30:02] DEBUG[6679] features.c: Removing dialed interfaces
datastore on Khomp/B1C0-0.0 since we're bridging
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Planning to masquerade
channel Khomp/B1C0-0.0 into the structure of
Local/21411615@test_outgoing-cb92;1
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Done planning to masquerade
channel Khomp/B1C0-0.0 into the structure of
Local/21411615@test_outgoing-cb92;1
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
write format slin
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
read format slin
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Putting channel
Khomp/B1C0-0.0 in slin/slin formats
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Done Masquerading
Khomp/B1C0-0.0 (6)
[2012-08-01 14:30:02] DEBUG[6679] chan_local.c: Not posting to
'Local/21411615@test_outgoing-cb92;2' queue since already masqueraded out*
***
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Bridge stops because we're
zombie or need a soft hangup: c0=Local/21411615@test_outgoing-cb92;2,
c1=Local/21411615@test_outgoing-cb92;1ZOMBIE, flags: No,Yes,Yes,Yes
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Bridge stops bridging
channels Local/21411615
Re: [asterisk-users] Problem with callfile and CDR
Ok. But the second leg is not recording the cdr. Is being generated only
the first leg of the cdr.
Regards.
Rodrigo Lang.
2012/8/1 Danny Nicholas da...@debsinc.com
Not a “bug” but a “feature”; when you use the local channel, the CDR is
recorded “incorrectly” because you are doing a 2-leg call.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, August 01, 2012 1:31 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Problem with callfile and CDR
** **
Hey, with the SIP works fine. Good tip.
** **
But is this a bug with Local?
** **
** **
Thanks!
Rodrigo Lang.
** **
** **
2012/8/1 Danny Nicholas da...@debsinc.com
Just a WAG, but could the “local” channel be causing some kind of
problem? Perhaps if you changed local to SIP or DAHDI?
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, August 01, 2012 12:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Problem with callfile and CDR
Good afternoon list.
I am experiencing a problem with the CDR and callfiles. What is happening
is this: When generating a call with a callfile, everything works
perfectly, but the CDR is recorded in the table when they answer the call
destination. The field disposition is being recorded correctly, but the
duration field is marked with the ring time and billsec is marked with 0.
This just happens to connections through callfiles. Yes, the call is
working usually remains. I did several tests with durations from seconds to
20 minutes.
I tested in two servers. With an Asterisk 1.4.44 (debian 5 64bits), and
another with Asterisk 1.8.14.0 (debian 6 64bits). In both occurs the same
problem. AND I tried using ForkCDR and ResetCDR and both did not help.
I'm doing something wrong? Has anyone experienced something similar? Any
tips?
*The callfile:*
Channel: local/21411615@test_outgoing
CallerID: ELCO Test 123456789
MaxRetries: 1
RetryTime: 30
WaitTime: 25
Context: test_ivr
Extension: 21411615
Priority: 1
AlwaysDelete: Yes
Archive: Yes
*The extensions.conf*
[test_outgoingsaida]
exten = _X.,1,Dial(khomp/gpstn/${EXTEN},120,Ttr)
exten = _X.,2,Hangup()
[test_ivr]
exten = _X.,1,Answer()
exten = _X.,n,Wait(20)
exten = _X.,n,Hangup()
*Example, console:*
*Log first channel:*
[2012-08-01 14:29:44] DEBUG[6679] pbx.c: Launching 'Dial'
[2012-08-01 14:29:44] VERBOSE[6679] pbx.c: -- Executing
[21411615@test_outgoing:1] Dial(Local/21411615@test_outgoing-cb92;2,
khomp/gpstn/21411615,120,Ttr) in new stack
[2012-08-01 14:29:44] DEBUG[6679] devicestate.c: device 'Khomp/B1C0' state
'1'
[2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Can't find native
functions for channel 'Khomp/B1C0-0.0'
[2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Seeded SDP of
'Khomp/B1C0-0.0' with that of 'Local/21411615@test_outgoing-cb92;2'
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDTIME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
ANSWEREDTIME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDPEERNAME.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALEDPEERNUMBER.
[2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable
DIALSTATUS.
[2012-08-01 14:29:44] VERBOSE[6679] app_dial.c: -- Called
khomp/gpstn/21411615
[2012-08-01 14:29:52] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0 is
ringing
[2012-08-01 14:29:53] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0 is
making progress passing it to Local/21411615@test_outgoing-cb92;2
[2012-08-01 14:30:02] VERBOSE[6679] app_dial.c: -- Khomp/B1C0-0.0
answered Local/21411615@test_outgoing-cb92;2
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
write format slin
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
read format slin
[2012-08-01 14:30:02] DEBUG[6679] features.c: Removing dialed interfaces
datastore on Khomp/B1C0-0.0 since we're bridging
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Planning to masquerade
channel Khomp/B1C0-0.0 into the structure of
Local/21411615@test_outgoing-cb92;1
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Done planning to masquerade
channel Khomp/B1C0-0.0 into the structure of
Local/21411615@test_outgoing-cb92;1
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp/B1C0-0.0 to
write format slin
[2012-08-01 14:30:02] DEBUG[6679] channel.c: Set channel Khomp
Re: [asterisk-users] Problem with callfile and CDR
Ow, thanks. Solve the issue! Adding /n at the end it worked correctly. Example: Channel: Local/21411615@test_outgoing/n Thanks again! Best regards, Rodrigo Lang. 2012/8/1 isr...@gmail.com add a /n at the end of the local channel -Original Message- From: Rodrigo Lang rodrigoferreiral...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 1 Aug 2012 15:53:44 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with callfile and CDR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is rsrvd and does not turn off
Hi Kevin. Thanks for the reply, can probably be just that. I'll contact the support that made the driver. In fact I sent the email to see if anyone has gone through a similar situation and give tips. Thank you, Rodrigo Lang. 2012/7/19 Kevin P. Fleming kpflem...@digium.com On 07/19/2012 03:49 PM, Rodrigo Lang wrote: I tried to shut down the channels with the command channel request hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing happens. I can only release these channels when I restart asterisk. You will probably need to ask the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is rsrvd and does not turn off
I checked with the plate holder and really is a driver error. Thanks, Rodrigo Lang. 2012/7/20 Rodrigo Lang rodrigoferreiral...@gmail.com Hi Kevin. Thanks for the reply, can probably be just that. I'll contact the support that made the driver. In fact I sent the email to see if anyone has gone through a similar situation and give tips. Thank you, Rodrigo Lang. 2012/7/19 Kevin P. Fleming kpflem...@digium.com On 07/19/2012 03:49 PM, Rodrigo Lang wrote: I tried to shut down the channels with the command channel request hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing happens. I can only release these channels when I restart asterisk. You will probably need to ask the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang -- Rodrigo Lang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel is rsrvd and does not turn off
Hi list.
I have Asterisk installed on a Debian 1.8 6 64-bit.
What happens is the following, some channels are not being hangup properly.
They run the hangup in dialplan, but the output of the command core show
channels shows several channels with status rsrvd. Checking the server's
memory, the top command shows multiple processes and stopped using the
Asterisk server memory.
I tried to shut down the channels with the command channel request hangup
Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing
happens. I can only release these channels when I restart asterisk.
Can one imagine what might be happening?
*Output of command core show channels:*
*
*
charger*CLI core show channels
Channel Location State Application(Data)
Khomp_SMS/B0C2-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp_SMS/B0C1-0 s@khomp-sms:1Rsrvd (None)
Khomp/B0C1-0.0 (None) Up AppDial((Outgoing Line))
SIP/9549-0f2cs@macro-dialout-trun Up
Dial(Khomp/*gebs/99451060,300,
13 active channels
1 active call
4884 calls processed
*Here's the context:*
context khomp-sms {
s = {
Noop(Mensagem recebida);
Noop(Tipo de mensagem: ${KSmsType});
if( ${KSmsType} = confirm ) {
Noop(Mensagem de confirmacao de entrega);
Agi(confirmation.py);
if( ${AGISTATUS} = SUCCESS) {
Log(NOTICE,AGI executado com sucesso);
} else {
Log(WARNING,Problema ao executar AGI - Status:
${AGISTATUS});
}
}
Hangup();
}
h = { Hangup(); }
}
*Follow the logs of the channel:*
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:1]
NoOp(Khomp_SMS/B0C0-0, Mensagem recebida) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:2]
NoOp(Khomp_SMS/B0C0-0, Tipo de mensagem: confirm) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:3]
GotoIf(Khomp_SMS/B0C0-0, 1?4:11) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Goto (khomp-sms,s,4)
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:4]
NoOp(Khomp_SMS/B0C0-0, Mensagem de confirmacao de entrega) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:5]
AGI(Khomp_SMS/B0C0-0, confirmation.py) in new stack
[2012-07-19 10:15:26] VERBOSE[554] res_agi.c: -- Khomp_SMS/B0C0-0AGI
Script confirmation.py completed, returning 0
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:6]
GotoIf(Khomp_SMS/B0C0-0, 1?7:9) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Goto (khomp-sms,s,7)
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:7]
Log(Khomp_SMS/B0C0-0, NOTICE,AGI executado com sucesso) in new stack
[2012-07-19 10:15:26] NOTICE[554] Ext. s: AGI executado com sucesso
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:8]
Goto(Khomp_SMS/B0C0-0, 10) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Goto (khomp-sms,s,10)
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:10]
NoOp(Khomp_SMS/B0C0-0, Finish if_if_khomp-sms_335_336) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:11]
NoOp(Khomp_SMS/B0C0-0, Finish if_khomp-sms_335) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [s@khomp-sms:12]
Hangup(Khomp_SMS/B0C0-0, ) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: == Spawn extension (khomp-sms,
s, 12) exited non-zero on 'Khomp_SMS/B0C0-0'
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: -- Executing [h@khomp-sms:1]
Hangup(Khomp_SMS/B0C0-0, ) in new stack
[2012-07-19 10:15:26] VERBOSE[554] pbx.c: == Spawn extension (khomp-sms,
h, 1) exited non-zero on 'Khomp_SMS/B0C0-0'
[2012-07-19 12:45:23] VERBOSE[554] asterisk.c: -- Remote UNIX
connection disconnected
I appreciate any help!
Best regards
--
Rodrigo Lang
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Re: [asterisk-users] extensions.lua with luasql.mysql.
Thanks a lot!
Best regards,
Rodrigo Lang.
2011/2/28 Borin katerin.bo...@gmail.com
Hi
try this pls
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18671
it did help to me
On Thu, Feb 24, 2011 at 9:31 PM, Rodrigo Lang
rodrigoferreiral...@gmail.com wrote:
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm not getting to run. When I reload the module
pbx_lua.so the following error appears:
[Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua
extension: error loading module 'luasql.mysql' from file
'/usr/lib/lua/5.1/luasql/mysql.so':
/usr/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_getfield
stack traceback:
[C]: ?
[C]: in function 'require'
[string extensions.lua]:205: in function [string
extensions.lua]:204
I tested my script with a file.lua and works ok and the extensions.lua
works fine too. My extensions.lua:
extensions = {
luatest = {
[302] = function()
require(luasql.mysql)
app.Answer()
app.Log(NOTICE, Trying to connect in MySQL)
app.Wait(2)
env = assert(luasql.mysql())
sql = assert
(env:connect(asterisk_teste,root,*,localhost,3306))
sel = sql:execute('SELECT * FROM cdr;')
sel:fetch(Fetcharray)
app.Noop(Fetcharray[1])
end;
h = function()
app.Hangup()
end;
};
}
Does anyone know what is happening?
Thansk in advance,
--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
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--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
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[asterisk-users] extensions.lua with luasql.mysql.
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm not getting to run. When I reload the module
pbx_lua.so the following error appears:
[Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua
extension: error loading module 'luasql.mysql' from file
'/usr/lib/lua/5.1/luasql/mysql.so':
/usr/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_getfield
stack traceback:
[C]: ?
[C]: in function 'require'
[string extensions.lua]:205: in function [string
extensions.lua]:204
I tested my script with a file.lua and works ok and the extensions.lua works
fine too. My extensions.lua:
extensions = {
luatest = {
[302] = function()
require(luasql.mysql)
app.Answer()
app.Log(NOTICE, Trying to connect in MySQL)
app.Wait(2)
env = assert(luasql.mysql())
sql = assert
(env:connect(asterisk_teste,root,*,localhost,3306))
sel = sql:execute('SELECT * FROM cdr;')
sel:fetch(Fetcharray)
app.Noop(Fetcharray[1])
end;
h = function()
app.Hangup()
end;
};
}
Does anyone know what is happening?
Thansk in advance,
--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
--
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[asterisk-users] Realtime MySQL - Asterisk 1.8.2
Hi to all. I make some tests with Asterisk 1.8.2 in Realtime. But i have one problem, the asterisk don't connect in the base and show this message: [Feb 17 11:18:01] WARNING[19061]: res_config_mysql.c:441 realtime_multi_mysql: MySQL RealTime: Invalid database specified: 'asterisk_teste' (check res_mysql.conf) I checked the asterisk config file (res_mysql.conf) and the configuration is ok. My configuration of table and extconfig.conf is the same of the version 1.6.0. The cdr use the same base, same user/pass, and his save the registers ok. This is happening to the queues, queues_members, muscionhold and queue_log in Realtime (That's is all the modules i use in Realtime). I tested the base and the MySQL and is working ok. I appreciate in advance any help, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime MySQL - Asterisk 1.8.2
2011/2/17 Ishfaq Malik i...@pack-net.co.uk On Thu, 2011-02-17 at 11:28 -0200, Rodrigo Lang wrote: Hi to all. I make some tests with Asterisk 1.8.2 in Realtime. But i have one problem, the asterisk don't connect in the base and show this message: [Feb 17 11:18:01] WARNING[19061]: res_config_mysql.c:441 realtime_multi_mysql: MySQL RealTime: Invalid database specified: 'asterisk_teste' (check res_mysql.conf) I checked the asterisk config file (res_mysql.conf) and the configuration is ok. My configuration of table and extconfig.conf is the same of the version 1.6.0. The cdr use the same base, same user/pass, and his save the registers ok. This is happening to the queues, queues_members, muscionhold and queue_log in Realtime (That's is all the modules i use in Realtime). I tested the base and the MySQL and is working ok. I appreciate in advance any help, -- Rodrigo Lang Opening your mind - Just another Open Source site The res_mysql.conf format changed from 1.6 to 1.8 The config now goes in in the following format [db-name] dbhost = x.x.x.x dbname = db-name dbuser = db-user dbpass = db-pass dbport = 3306 Is this how yours is set up? Thanks a lot, i solved the problem. First, the WARNING about 'res_mysql.conf', but the file now is 'res_config_mysql.conf'. Second, i changed the '[generals]' to the name of my database. Worked ok! Thanks again! -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR with unix time.
2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. Hi. I tested in the version 1.6.0 and works fine. Thanks a lot. Best regards, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR with unix time.
Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject an incoming call using AMI ?
Hi. You see the comando Hangup in the AMI? Best regards, Rodrigo Lang. 2011/1/10 Olivier oza_4...@yahoo.fr Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means let another agent answer this call (it doesn't mean end this call). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup the agent call leg and then redirect the caller back to the appropriate queue, hoping the caller wouldn't be once again forwarded to the busy agent. Which way to implement this would you suggest or recommend ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reload queue on the fly?
Try: module reload app_queue.so 2010/12/28 Денис Давыдов dyna...@gmail.com Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reload queue on the fly?
Try modify the queues.conf to this: [office] strategy = linear timeout = 10 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 At, Rodrigo Lang. 2010/12/28 Давыдов Денис dyna...@gmail.com The same result. Colleague did remotely (in his words): `queue reload all office' - and it works for me. This is very strange why my variant didn't work :( On 12/28/2010 03:54 PM, Rodrigo Lang wrote: Try: module reload app_queue.so 2010/12/28 Денис Давыдов dyna...@gmail.com Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- С уважением, Денис Давыдов -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debug messages.
Good morning to all. In my Asterisk console i have a lot of this messages: [Dec 14 10:50:52] DEBUG[12790]: audiohook.c:215 audiohook_read_frame_both: Read factory 0x8afae68 and write factory 0x8afb884 both fail to provide 160 samples [Dec 14 10:50:52] DEBUG[12790]: audiohook.c:221 audiohook_read_frame_both: Write factory 0x8afb884 was pretty quick last time, waiting for them. Someone can tell me what this messages mean? Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abandon events in cdr
Sorry, of course cdr.conf not queues.conf. marcus Am 01.12.2010 19:16 schrieb marcus rothe sync...@googlemail.com: Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? Regards Marcus Thanks very much, I include the line unansweredy=yes in the cdr.conf and solve the problem. Thanks again! -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ I messed up in the records, did not fix the problem. The calls that are going to leave the queue and still are no records in the cdr! The output of the command show status cdr: AST * CLI cdr status show CDR logging: enabled CDR mode: simple CDR output unanswered calls: no CDR registered backend: mysql The option of unanswered calls is to no, but is cdr.conf configre to yes. Look: cat / etc / asterisk / cdr.conf [General] enable = yes unansweredy = yes safeshutdown = yes endbeforehexten = yes [Mysql] host = localhost Thanks again, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abandon events in cdr
No, i am standing absolutely at the beginning. I think the table structure should be googleable. But i haven 't found an automatism to dump the queuelog flatfile into a database table. Found a perl script but it doesn' t work for me. Am 03.12.2010 19:29 schrieb Rodrigo Lang rodrigoferreiral...@gmail.com : 2010/12/3 marcus rothe sync...@googlemail.com That sounds good. Rodrigo, allow me one question. I'm not very familar with databases but have the need to report out of the queuelog. Have you got a hint for me how to export the queueslog file into a database table? Thanks in advance, Regards Marcus Hi. Assuming you use Debian and have all your repositories ok, do: Install mysql Database: aptitude -y install mysql-server libmysqlclient15-dev mysql-client Install php5 to run my script: aptitude -y install php5 php5-cli php5-cgi php5-mysql To create the MySQL table I use this script [1], I pasted in the pastebin now. This script creates the CDR and Queue_log tables. To convert your cvs queue_log for MySQL, use this script [2]. I did it now and have tested, it worked for me. Make a backup of queue_log before use. [1] http://pastebin.com/2v5UPg3Q [2] http://pastebin.com/TCJHkPXP Any questions just ask. If you find an error in php script just let me know, then I stand corrected. And please answer in the list. So when someone needs the same procedure like you needed now, he can find in list history. Best regards, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi on Realtime.
There is no specific Realtime database for chan_dahdi (that I know if). You can store the configuration using Realtime Static using the new chan_dahdi.conf notation without any problems. The only problem with Realtime Static is that you cannot use the text file, you need to load everything from the database. Another possibility would be to use an #exec from chan_dahdi.conf to extract the channel configuration from the database. Thanks for the reply Carlos. You have the model of the tables for chan_dahdi in static mode? This quite difficult to find on the internet ... And you know if the generals can also be included in a static way? Thanks again At, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi on Realtime.
Good morning list. I wonder if I can put files and chan_dahdi dahdi_channels in real time. Not the generals but the channels. Thanks, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reasons of OriginateResponse
Good morning everyone. I wonder where I can find a list of the reasons the event OriginateResponse. I found this list [1]. But in my Asterisk has other reasons too. [1] 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Queue_log and CDR.
Good afternoon list. I am facing a problem with the CDR and Queue_log tables (MySQL). The ABANDON events is being saved correctly in queue_log, but in the table CDR is not saving the registry of such abandoned calls. Apparently the CDR table is functioning normally, I have several records of links in it. From what I noticed, is only the events abandonment that are malfunctioning. With this SELECT [1] I can pick up the records on other servers without problems. With this another SELECT [2], I get the events Abaddon normally. With this other [3] I can get all the channels that joined the queue with no problems. Both tables are recording normally and Asterisk has no errors in the logs. Only happens with the event Abandon. [1] SELECT from_unixtime(ql.time),ql.callid,ql.event FROM cdr c,queue_log ql WHERE c.uniqueid = ql.callid AND ql.event = 'ABANDON'; [2] SELECT * FROM queue_log WHERE event = 'ABANDON'; [3] SELECT from_unixtime(ql.time),ql.callid,ql.event FROM cdr c,queue_log ql WHERE c.uniqueid = ql.callid AND ql.event = 'ENTERQUEUE'; Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abandon events in cdr
Sorry, of course cdr.conf not queues.conf. marcus Am 01.12.2010 19:16 schrieb marcus rothe sync...@googlemail.com: Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? Regards Marcus Thanks very much, I include the line unansweredy=yes in the cdr.conf and solve the problem. Thanks again! -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate Response.
Hi to all. I am conducting several tests with the Asterisk manager and I noticed something that I believe to be a problem. When I generate a call with the Action Originate with the Async option true, the event OriginateResponse returns normally. But when I generate a call in the same way, without the Async option, the event OriginateResponse does not come. Is this a bug? It was fixed in some version? I use Asterisk version 1.6.0.28 Thanks in advance. -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make call in AMI.
[Nov 19 16:49:56] NOTICE[23371]: chan_local.c:655 local_alloc: No such
extension/context 04191028...@intermovel creating local channel
can you display interMovel context ?
Is there any entry matching 0419102889 in interMovel context ?
Oh! Like I'm stupid. This basic detail went totally unnoticed. Thanks for
the reply and sorry for my fault. The context:
context interMovel {
_00XX[7-9]XXX = { saidaGSM(${EXTEN:1}); }
h = { hangupGlobal(); }
}
It worked properly when I added one more zero. Thanks again.
At,
--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
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[asterisk-users] Make call in AMI.
Good afternoon list. I need to make calls via AMI, but I need to leave the links in their respective contexts, to mobile phone calls by leaving out the context of mobile and so on. Already configured the settings that way, but I do not like the the Action Originate do it. I tried several ways, none successfully. What came closer to work the way I need is this: action: originate channel: Local/04191028...@intermovel context: returnCall extension: *10198 priority: 1 async: true interMovel is my context. But the answer on the Asterisk console was this: [Nov 19 16:49:56] NOTICE[23371]: chan_local.c:655 local_alloc: No such extension/context 04191028...@intermovel creating local channel [Nov 19 16:49:56] NOTICE[23371]: channel.c:3854 __ast_request_and_dial: Unable to request channel Local/04191028...@intermovel [Nov 19 16:49:56] ERROR[3843]: pbx.c:8396 device_state_cb: Received invalid event that had no device IE [Nov 19 16:49:56] ERROR[3843]: app_queue.c:862 device_state_cb: Received invalid event that had no device IE I need to do the links go out into different channels according to what is configured in the dialplan and dynamically. I can make a call by calling the channel normally, thus: action: originate channel: DAHDI/g1/04191028897 context: returnCall extension: *10198 priority: 1 async: true Does anyone have any idea how to do? Thank you in advance, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with AMI
Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-6fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid: 1289414204.29705 This guy is ok. But sometimes the event come like this: Event: Newstate Privilege: call,all Channel: SIP/17-6fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: CallerIDName: Uniqueid: 1289414204.29705 And sometimes these events don't come. I think this is a bug, correctly? My Asterisk version is the 1.6.0.28. I use ATA Linksys in the extensions and this problem happen in every hardware's (khomp, xorcom) and in the SIP protocol to. Note: Every teste made is from the same number (my cell) and to same extension of destiny. Thanks a lot to all. -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get the Uniqueid of Action Originate in the AMI
Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a lots, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
Thanks a lot to all for the responses. I begin to use the event OriginateResponse, it's what i need. Thanks again. Best regards, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call using password
What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Hey Flavio, you are a brazilian, it's right? In my blog i have three exemples of how to make call using a password, but is in pt_BR. First exemple [1] use a phpAgi and PostgreSQL. Second exemple [2] is more simple, use a AstDB and Authenticate(). The third [3] is a lock feature. [1] http://rodrigorecipes.blogspot.com/2010/05/upgrade-na-facilidade-ligacao-por-login.html [2] http://rodrigorecipes.blogspot.com/2010/05/proteja-seu-bolso-com-aplicacao_05.html [3] http://rodrigorecipes.blogspot.com/2010/05/proteja-seu-bolso-com-aplicacao.html Best regards, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and SIP a Provider in Brazil
I have sent an e-mail to this list (awaiting moderator approval by the size) talking about some difficult to make calls with a SIP Provider in Brazil. I'm new at this list and have no sure if I have posted my question in the right place. If this is not the channel to make this kind of question about this issue, I'm sorry but want to ask if anyone can indicate the correct place . If you are brazilian, enter in brasilian comunity of Asterisk [1]. Ther you will have the best information about Voip providers in Brazil. The people of AsteriskBrasil have a lots of experience with the providers in Brazil. [1] http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil Att, -- Rodrigo Lang, Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CALLERPRES() with Queue
Good afternoon list, I'm having a problem using the function CALLERPRES() when connection to a Queue(). When I call an extension, before the Dial (), I select the function CALLERPRES () as unavailable to link the extension comes as anonymous. But if I call a queue before the Queue (), I select the function CALLERPRES() as unavailable, but the identification appears normal. Is it a problem or configuration? Someone can have for that? Regards, -- Rodrigo Lang http://rodrigorecipes.blogspot.com/http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI Originate
3 miliseconds... 2010/10/1 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Cropp *Sent:* Friday, October 01, 2010 3:50 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] AMI Originate snip Timeout: 3 3 seconds to answer the phone? You never get coffee? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang http://rodrigorecipes.blogspot.com/http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
Use the command aptitude install tofrodos to install dos2unix. This command get the file and clear the ^M. Regards, Rodrigo Lang. 2010/8/25 Zeeshan Zakaria zisha...@gmail.com Actually I have found the problem, and leanred some new stuff along with it. Apparently all Linux files have a mime type information stored in them, which can be checked using command: file -i filename For my extensions.ael, which I copied from a different server, the mime type is 'text/x-c' whereas all the other files have mime type 'text/plain'. Now if I create a new file extemsions.maelstrom on this machine which is by default 'text/plain', there are no errors on doing 'ael reload', however using extensions.maelstrom with mime type 'text/x-c' gives errors, though the code works fine. How it got mime type 'text/x-c' on the other machine, Vim which I use there assigned it this mime type. I'll have to fix it there. Now I am trying to figure out how to convert between mime types. A simpler solution is to just copy text to a new file, but would be nice to do a proper conversion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 12:01 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: -- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On B... *Sent:* Wednesday, August 25, 2010 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s): That's what I understood too from this one and probably only related google search result, but even ... Is there any chance that these files were edited on a Windows machine and then copied back to the Asterisk boxes? That is, are there some nefarious ^m characters hiding in there? Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang http://rodrigorecipes.blogspot.com/http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi, thanks a lot by the answers. But without the application Answer() the problem remains. Realized over a battery of tests and refined the problem. Follows: A = External link that came with my Voip number. B = Operator. C = The extent to which A want to speak. A called my number and B answer. If B try to transfer with blindxfer (#) to C works fine. But if B try to transfer with atxfer (*2) he can talk to C, only when B hangs up to complete the transfer begins to generate those warnings on the cli. After the transfer using C atxfer not hear A, but A hears C. I believe it has become clearer now. And as he said, with any codec, and only when the person connects to my VoIP trunks. I did the test with the analogue trunks and atxfer worked normal. Thanks, Rodrigo Lang. 2010/7/20 Stefan Schmidt s...@sil.at Rodrigo Lang schrieb: Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning: [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang. hello rodrigo, this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) There is a patch for asterisk 1.6.2.9 but its only a single row so you could easy find the position in app_dial.c to patch it by your own. the problem only occurs when you use answer in your dialplan. without an answer this wont happen. best regards. steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP
Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning: [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
This is the exit of core show version: Asterisk 1.6.0.28 built by root @ AST on a i686 running Linux on 2010-06-28 12:21:24 UTC Obg, Rodrigo Lang. 2010/7/20 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. I *think* there is/was an entry in the bug tracker on this. You might want to search https://issues.asterisk.org (also look for RTP issues with SSRC) and in the meanwhile you could reveal which version of Asterisk you are using. :) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recordings in the bank.
Hello list. I've been researching if there is a way of putting the recordings of Mixmonitor in database (PostgreSQL or MySQL) in an automated way. I've read that the native has voicemail in Asterisk via ODBC. And for the MixMonitor has some way? Someone on the list have it implemented? Thanks, Rodrigo Lang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But
after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
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Re: [asterisk-users] Return agi script.
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return
1).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 1:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Return agi script.
Hi Danny. I solve the problem. I put exit (return); where return is equal
to ${AGISTATUS} text. Example:
exit(SUCCESS);
exit(FAILURE);
exit(HANGUP);
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
text string, one of SUCCESS | FAILURE | NOTFOUND |
HANGUP
:D
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Can you post the script?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 2:09 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Return agi script.
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return
1).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 1:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
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[asterisk-users] Find a way to block brute force attacks.
Hello list. I'm trying to find a way to block any ip that tries to login more than three times with the wrong password and try to log in three different extensions. For I have suffered some brute force attacks on my asterisk in the morning period. The idea would be: Any ip with three attempts without success to log into an extension is blocked. Is there any way to accomplish this directly by the asterisk? Or is there some kind of asterisk spit this information via the AMI? I was wondering to make a Java program to listen to the AMI and create a rule in iptables for ip in specific. Does anyone have any suggestions? Thanks, Rodrigo Lang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Find a way to block brute force attacks.
Good afternoon. Thanks to everyone for answers. What I find strange is the asterisk does not have any native tool for him to SIP server security. Here's an example of the syslog messages from asterisk: [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password From what I told there is around twenty thousand records that at one time. And at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson kwat...@geniusgroupltd.com Hi, you can use fail2ban http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asteriskhttp://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk Which works well, when a pattern is found in a log file it addes in an iptables rules to block the traffic for a period. On debian you can apt-get install fail2ban and on centos/redhat yum -i fail2ban Thanks Kenny - Original Message - From: Gareth Blades list-aster...@skycomuk.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 June, 2010 4:12:42 PM Subject: Re: [asterisk-users] Find a way to block brute force attacks. Rodrigo Lang wrote: Hello list. I'm trying to find a way to block any ip that tries to login more than three times with the wrong password and try to log in three different extensions. For I have suffered some brute force attacks on my asterisk in the morning period. The idea would be: Any ip with three attempts without success to log into an extension is blocked. Is there any way to accomplish this directly by the asterisk? Or is there some kind of asterisk spit this information via the AMI? I was wondering to make a Java program to listen to the AMI and create a rule in iptables for ip in specific. Does anyone have any suggestions? Thanks, Rodrigo Lang. Does asterisk log the failed attempts to a file? If so then you could use sshblack to monitor the file for incorrect logins. It will add firewalls rules to a custom iptables chain based on various criteria. You can then point incoming SIP connections through this chain so offenders will be forewalled for a specific amount of time. http://www.pettingers.org/code/sshblack.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Find a way to block brute force attacks.
Thanks again. But it was a question pending. It's possible AMI show failure resgisters and wrong password? Because I already have a Java program for AMI and a few lines of modification would solve my problem if asterisk sends the information to the AMI. Thanks, Rodrigo Lang. 2010/6/29 Andrew Latham lath...@gmail.com Please start here http://www.spamhaus.org/drop/ with your BGP routes Then move up to log parsing. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Jun 29, 2010 at 1:38 PM, Zeeshan Zakaria zisha...@gmail.com wrote: If I didn't have fail2ban, I would have way over 20k of these entries in my asterisk log. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-29 1:36 PM, Rodrigo Lang rodrigoferreiral...@gmail.com wrote: Good afternoon. Thanks to everyone for answers. What I find strange is the asterisk does not have any native tool for him to SIP server security. Here's an example of the syslog messages from asterisk: [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password [Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '213 sip:2...@my_extern_ip' failed for '116 .124.128.82 '- Wrong password From what I told there is around twenty thousand records that at one time. And at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson kwat...@geniusgroupltd.com Hi, you can use fail2ban http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asteri.http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asteri. .. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
