Re: [asterisk-users] Cisco 3950 ip phone

[Gokan Atmaca]

> The phone does work, you do need to TFTP the configuration files to the phone 
> though.  Doesn't look like custom firmware is required.

very thanks...

the following ?
cisco ip phone:  8905 , 8950, 8450



On Fri, Apr 12, 2019 at 4:32 PM Chris Knipe  wrote:
>
> Hi,
>
> https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981
>
> The phone does work, you do need to TFTP the configuration files to the phone 
> though.  Doesn't look like custom firmware is required.
>
> --
> Chris.
>
>
> On Fri, Apr 12, 2019 at 3:29 PM Antony Stone 
>  wrote:
>>
>> On Friday 12 April 2019 at 15:24:27, Gokan Atmaca wrote:
>>
>> > > Please give us a link to a datasheet for that device.
>> >
>> > Hello
>> >
>> > https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/u
>> > nified-sip-phone-3905/data_sheet_c78-651588.html
>>
>> Ah - that explains why I couldn't find a model 3950 phone :)
>>
>> Well, it uses SIP, so it should work as well as any Cisco phone does on
>> Asterisk.
>>
>>
>> Antony.
>>
>> > On Fri, Apr 12, 2019 at 3:58 PM Antony Stone wrote:
>> > > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote:
>> > > > Hello
>> > > >
>> > > > Can I use Cisco 3950 on Asterisk ?
>> > >
>> > > Please give us a link to a datasheet for that device.
>>
>> --
>> BASIC is to computer languages what Roman numerals are to arithmetic.
>>
>>Please reply to the list;
>>  please *don't* CC 
>> me.
>>
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>>
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>
>
>
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> Regards,
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Re: [asterisk-users] Cisco 3950 ip phone

[Gokan Atmaca]

> Please give us a link to a datasheet for that device.

Hello

https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-sip-phone-3905/data_sheet_c78-651588.html

On Fri, Apr 12, 2019 at 3:58 PM Antony Stone
 wrote:
>
> On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote:
>
> > Hello
> >
> > Can I use Cisco 3950 on Asterisk ?
>
> Please give us a link to a datasheet for that device.
>
> Regards,
>
>
> Antony
>
> --
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[asterisk-users] Cisco 3950 ip phone

[Gokan Atmaca]

Hello

Can I use Cisco 3950 on Asterisk ?

Thanks.

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Re: [asterisk-users] Res_Srtp

[Gokan Atmaca]

> If compiling, you'll need to install the development library.  Under
> Debian it is libsrtp0-dev

Hello
I installed the package (libsrtp0-dev) and re-compiled it. currently
sees the module.

sip0*CLI> module show like res_srtp.so
Module Description
 Use Count  Status  Support Level
res_srtp.soSecure RTP (SRTP)
 0  Running  core
1 modules loaded


Thanks.


On Sun, Mar 31, 2019 at 4:36 PM Doug Lytle  wrote:
>
> On 3/31/19 8:21 AM, Gokan Atmaca wrote:
> > Hello
> >
> > The "res_srtp" module does not appear. How do I install it?
> >
>
> Are you compiling or installing from packages?
>
> If compiling, you'll need to install the development library.  Under
> Debian it is libsrtp0-dev
>
> Doug
>
>
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[asterisk-users] Res_Srtp

[Gokan Atmaca]

Hello

The "res_srtp" module does not appear. How do I install it?

Thanks.

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[asterisk-users] IVR Loop

[Gokan Atmaca]

Hello

IVR call is coming. I want 8 (digit) in the loop. How can I do that ?

Current confi:
[ivr1]
exten=>_,1,answer()
exten=>_,n,background(/var/lib/asterisk/ivr/ob)
exten=>_,n,WaitExten(10)
exten=>_,n,Dial(${OPERATOR})
exten=>i,1,Dial(${OPERATOR})
exten=>t,1,Dial(${OPERATOR})
exten=>1,1,dial(SIP/6001)
exten=>2,1,dial(SIP/6002)
exten=>3,1,dial(SIP/6003)


Thanks.

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Re: [asterisk-users] internal call record

[Gokan Atmaca]

> Well, firstly, why are you using "_6001"?  The leading underscore says "this 
> is
> going to be a pattern match", but you then follow with a specific number, not 
> a
> pattern.  You might as well write just "6001" without the underscore, which is
> clearer to you and more efficient for Asterisk.


Hello

The total number of internal medicine is 6000-6020. When I do 6XXX
getting error. I did as follows. I think (*) there is a mistake
I am doing.


exten => 6XXX,1,NoOp()
exten => 6XXX,n,MixMonitor(${UNIQUEID}.wav,ab)
exten => 6XXX,n,Dial(SIP/${ARG2}/${ARG1},20)   (*)
exten => 6XXX,n,StopMixMonitor()
exten => 6XXX,n,Hangup()



On Sun, Mar 10, 2019 at 12:59 PM Antony Stone
 wrote:
>
> On Sunday 10 March 2019 at 10:46:24, Gokan Atmaca wrote:
>
> > Hello
> >
> > Mynum: 6001 , Othernum: 6002.
> >
> >
> > I can record as follows.
>
> So, are you saying that this is the part which does work?
>
> We need to see the part which doesn't work as well, otherwise we have no idea
> what to suggest you change.
>
> > But I do not enter individual records for each internal required. I want to
> > do it more smoothly with a Macro.
>
> > exten => _6001,1,NoOp()
> > exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab)
> > exten => _6001,n,Dial(SIP/6001,20)
> > exten => _6001,n,StopMixMonitor()
> > exten => _6001,n,Hangup()
>
> Well, firstly, why are you using "_6001"?  The leading underscore says "this 
> is
> going to be a pattern match", but you then follow with a specific number, not 
> a
> pattern.  You might as well write just "6001" without the underscore, which is
> clearer to you and more efficient for Asterisk.
>
> Note: You don't need the StopMixMonitor() immediately preceding a Hangup() -
> it's done automatically when the call ends.
>
> Do all your internal extensions start with "600" (or maybe "60")?
>
> If so, have you tried the same dial plan but using "_600X" or "_60XX" as the
> extension match on every line (oh, and incidentally, you can say "same => n,"
> in place of "exten => _6000,n," on every line after the first one - this saves
> some typing, and also makes things a lot simpler if you decide you need to
> change the basic pattern match at some time in the future).
>
> > On Sat, Mar 9, 2019 at 6:50 PM Doug Lytle wrote:
> > > On Sat, Mar 9, 2019 at 4:25 PM Antony Stone wrote:
> > >
> > > a) work for recording incoming / outgoing calls
> > >
> > > b) do not work for recording internal calls
> > >
> > > then we might be able to give you a clue what's wrong.
> > >
> > > Hello
> > >
> > > For example: My phone number is 1000, the other's number is 1001. These
> > > numbers are in the same PBX (asterisk). I want 1000, 1001
> > >
> > >
> > > Gokan,
> > >
> > >
> > > Since you've said that outside calls can be recorded, but not inside
> > > calls; Antony requested that you show us your dialplan code for
> > > recordings that work.  This will give us an idea of what might be going
> > > wrong when trying to record inside calls.
> > >
> > > It would also be helpful to see your console output when things are not
> > > working.
> > >
> > > Doug
>
> --
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>
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>  please *don't* CC me.
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Re: [asterisk-users] internal call record

[Gokan Atmaca]

Hello

Mynum: 6001 , Othernum: 6002.


I can record as follows. But I do not enter individual records for each internal
required. I want to do it more smoothly with a Macro.

Thanks.

exten => _6001,1,NoOp()
exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab)
exten => _6001,n,Dial(SIP/6001,20)
exten => _6001,n,StopMixMonitor()
exten => _6001,n,Hangup()


On Sat, Mar 9, 2019 at 6:50 PM Doug Lytle  wrote:
>
> On 3/9/19 9:56 AM, Gokan Atmaca wrote:
>
> a) work for recording incoming / outgoing calls
>
> b) do not work for recording internal calls
>
> then we might be able to give you a clue what's wrong.
>
> Hello
>
> For example: My phone number is 1000, the other's number is 1001. These 
> numbers
> are in the same PBX (asterisk). I want 1000, 1001
>
>
> Gokan,
>
>
> Since you've said that outside calls can be recorded, but not inside calls; 
> Antony requested that you show us your dialplan code for recordings that 
> work.  This will give us an idea of what might be going wrong when trying to 
> record inside calls.
>
> It would also be helpful to see your console output when things are not 
> working.
>
> Doug
>
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Re: [asterisk-users] internal call record

[Gokan Atmaca]

> Show us the parts of your dial plan which:
>
> a) work for recording incoming / outgoing calls
>
> b) do not work for recording internal calls
>
> then we might be able to give you a clue what's wrong.

Hello

For example: My phone number is 1000, the other's number is 1001. These numbers
are in the same PBX (asterisk). I want 1000, 1001 can call to record
audio calls.

Thanks.

On Sat, Mar 9, 2019 at 4:25 PM Antony Stone
 wrote:
>
> On Saturday 09 March 2019 at 14:19:19, Gokan Atmaca wrote:
>
> > Hello
> >
> > How can I record voice between internalities? I can record voice in
> > incoming and outgoing calls, but I can't make it between the internal.
> > Would you support this?
>
> Show us the parts of your dial plan which:
>
> a) work for recording incoming / outgoing calls
>
> b) do not work for recording internal calls
>
> then we might be able to give you a clue what's wrong.
>
>
> Antony.
>
> --
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>
>  - Billy Connolly
>
>Please reply to the list;
>  please *don't* CC me.
>
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>
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[asterisk-users] internal call record

[Gokan Atmaca]

Hello

How can I record voice between internalities? I can record voice in
incoming and outgoing calls, but I can't make it between the internal.
Would you support this?

Thanks.

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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

> 1. What is the content of ${OPERATOR}?
>
> 2. What do you have for this connection in sip.conf?
>
> 3. What number/s have you been assigned by your upstream SIP provider?
>
> Antony.

Hello

The problem appeared in siptrunk. The problem is "insecure=very". This
"insecure=invite" improved.

Very thanks.

On Tue, Mar 5, 2019 at 7:29 PM Antony Stone
 wrote:
>
> On Tuesday 05 March 2019 at 17:22:16, Gokan Atmaca wrote:
>
> > > exten => _13XXX,1,dial(${OPERATOR},20)
>
> 1. What is the content of ${OPERATOR}?
>
> 2. What do you have for this connection in sip.conf?
>
> 3. What number/s have you been assigned by your upstream SIP provider?
>
> Antony.
>
> > On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle  wrote:
> > > On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > > > Asterisk can send calls, but I don't get a call. What could be the
> > > > problem?
> > > >
> > > > [from-siptrunk]
> > > > exten => 13XXX,1,dial(${OPERATOR},20)
> > >
> > > exten => _13XXX,1,dial(${OPERATOR},20)
> > >
> > > Doug
>
> --
> In science, one tries to tell people
> in such a way as to be understood by everyone
> something that no-one ever knew before.
>
> In poetry, it is the exact opposite.
>
>  - Paul Dirac
>
>Please reply to the list;
>  please *don't* CC me.
>
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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

> exten => _13XXX,1,dial(${OPERATOR},20)

Hello

"SIP/2.0 401 Unauthorized"  Unfortunately the negative. An asterisk
indicates a 404 error.



On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle  wrote:
>
> On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > Asterisk can send calls, but I don't get a call. What could be the problem?
> >
> > [from-siptrunk]
> > exten => 13XXX,1,dial(${OPERATOR},20)
> >
>
> exten => _13XXX,1,dial(${OPERATOR},20)
>
> Doug
>
> Doug
>
>
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[asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

Hello

Asterisk can send calls, but I don't get a call. What could be the problem?

[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)

Thanks.

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Re: [asterisk-users] Some SIP and IAX Asterisk unreachable after server restart

[Gokan Atmaca]

>
> Everything is working again if we restart the customers Asterisk.

Have you looked at the error logs?

On Mon, Feb 27, 2017 at 3:03 PM, Administrator TOOTAI  wrote:
> Hi all,
>
> we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 7.11
> (wheezy), the host OS being the same.
>
> Problem: when we restart the server (eg host + VM), all customers Asterisk
> connecting without a VPN (doesn't matter which Asterisk version) are no more
> reachable. Same for IAX users.
>
> On the host side, after restart, SIP packet are entering the host but NOT to
> the VM (tshark debug). In IAX, POKE packets sended by the VM are reaching
> the client Asterisk but no packet answer.
>
> Everything is working again if we restart the customers Asterisk.
>
> Any clue ?
>
> Regards
>
> --
> Daniel
>
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Re: [asterisk-users] Fail2ban

[Gokan Atmaca]

I solved the problem. "action.d/iptables-custom.conf"  include only udp.
service fail2ban restart

Thank you.

On Sun, Sep 13, 2015 at 9:17 PM, Andres <and...@telesip.net> wrote:
> On 9/13/15 11:16 AM, Gokan Atmaca wrote:
>>
>> Hello
>>
>> I'm using the Fail2ban.  I configuration below. I want to try to
>> prevent the continuous password. Fail2ban password that does not
>> prevent this form. (Asterisk 1.8 / Elastix interface)
>>
>> What could be the problem ?
>>
>> Asterisk log;
>> "Registration from '<sip:3...@sip.x.eu;transport=UDP>' failed for
>> 'x.x.x.x:32956' - Wrong password"
>
> Sometimes minor tweaks to the file are in order.  My suggestion is to use
> the fail2ban-regex utility to test the log file entry until it is detected.
> Just put the line generated by asterisk in a test file and then run the
> regex.
>
> # /usr/bin/fail2ban-regex -?
> Usage: /usr/bin/fail2ban-regex [OPTIONS]   [IGNOREREGEX]
>
> example:
>
> /usr/bin/fail2ban-regex testlogfile /etc/fail2ban/filter.d/asterisk.conf
>
>
>
>
>
>>
>>
>> Fail2ban asterisk filter;
>>
>> # Fail2Ban filter for asterisk authentication failures
>> #
>>
>> [INCLUDES]
>>
>> # Read common prefixes. If any customizations available -- read them from
>>
>> # common.local
>> before = common.conf
>>
>>
>> [Definition]
>>
>> _daemon = asterisk
>>
>> __pid_re = (?:\[\d+\])
>>
>> # All Asterisk log messages begin like this:
>> log_prefix= (?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>> \S+:\d*( in \w+:)?
>>
>> failregex = ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Registration
>> from '[^']*' failed for '(:\d+)?' - (Wrong
>> password|Username/auth name mismatch|No m$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>> not found in context 'de$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed to authenticate as '[^']*'$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>> for peer '[^']*' \(from \)$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed MD5 authentication for '[^']*' \([^)]+\)$
>>^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>> not found in context 'de$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed to authenticate as '[^']*'$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>> for peer '[^']*' \(from \)$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed MD5 authentication for '[^']*' \([^)]+\)$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Failed to
>> authenticate (user|device) [^@]+@\S*$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>> (?:handle_request_subscribe: )?Sending fake auth rejection for
>> (device|user) \d*<sip:[^@]+@>;tag=$
>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>>
>> SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="[\d-]+",S$
>>
>> ^(%(__prefix_line)s|\[\]\s*WARNING%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>> )Ext\. s: "Rejecting unknown SIP connection from "$
>>
>> ignoreregex =
>>
>>
>> # Author: Xavier Devlamynck / Daniel Black
>> #
>> # General log format - main/logger.c:ast_log
>> # Address format - ast_sockaddr_stringify
>> #
>> # First regex: channels/chan_sip.c
>> #
>> # main/logger.c:ast_log_vsyslog - "in {functionname}:" only occurs in s
>>
>
>
> --
> Technical Support
> http://www.cellroute.net
>
>
>
> --
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Re: [asterisk-users] Fail2ban

[Gokan Atmaca]

Another problem is too late to do the ban. The reason for this
yetmemse of CPU power. I'm simulating an attack. Of course, eating
CPU. One reason, now forbids. Abstracts must be strong if we are
eating our resources is a serious attack.

On Mon, Sep 14, 2015 at 9:14 AM, Gokan Atmaca <linux.go...@gmail.com> wrote:
> I solved the problem. "action.d/iptables-custom.conf"  include only udp.
> service fail2ban restart
>
> Thank you.
>
> On Sun, Sep 13, 2015 at 9:17 PM, Andres <and...@telesip.net> wrote:
>> On 9/13/15 11:16 AM, Gokan Atmaca wrote:
>>>
>>> Hello
>>>
>>> I'm using the Fail2ban.  I configuration below. I want to try to
>>> prevent the continuous password. Fail2ban password that does not
>>> prevent this form. (Asterisk 1.8 / Elastix interface)
>>>
>>> What could be the problem ?
>>>
>>> Asterisk log;
>>> "Registration from '<sip:3...@sip.x.eu;transport=UDP>' failed for
>>> 'x.x.x.x:32956' - Wrong password"
>>
>> Sometimes minor tweaks to the file are in order.  My suggestion is to use
>> the fail2ban-regex utility to test the log file entry until it is detected.
>> Just put the line generated by asterisk in a test file and then run the
>> regex.
>>
>> # /usr/bin/fail2ban-regex -?
>> Usage: /usr/bin/fail2ban-regex [OPTIONS]   [IGNOREREGEX]
>>
>> example:
>>
>> /usr/bin/fail2ban-regex testlogfile /etc/fail2ban/filter.d/asterisk.conf
>>
>>
>>
>>
>>
>>>
>>>
>>> Fail2ban asterisk filter;
>>>
>>> # Fail2Ban filter for asterisk authentication failures
>>> #
>>>
>>> [INCLUDES]
>>>
>>> # Read common prefixes. If any customizations available -- read them from
>>>
>>> # common.local
>>> before = common.conf
>>>
>>>
>>> [Definition]
>>>
>>> _daemon = asterisk
>>>
>>> __pid_re = (?:\[\d+\])
>>>
>>> # All Asterisk log messages begin like this:
>>> log_prefix= (?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>>> \S+:\d*( in \w+:)?
>>>
>>> failregex = ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Registration
>>> from '[^']*' failed for '(:\d+)?' - (Wrong
>>> password|Username/auth name mismatch|No m$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>>> not found in context 'de$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>>> failed to authenticate as '[^']*'$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>>> for peer '[^']*' \(from \)$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>>> failed MD5 authentication for '[^']*' \([^)]+\)$
>>>^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>>> not found in context 'de$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>>> failed to authenticate as '[^']*'$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>>> for peer '[^']*' \(from \)$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>>> failed MD5 authentication for '[^']*' \([^)]+\)$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Failed to
>>> authenticate (user|device) [^@]+@\S*$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>>> (?:handle_request_subscribe: )?Sending fake auth rejection for
>>> (device|user) \d*<sip:[^@]+@>;tag=$
>>>  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>>>
>>> SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="[\d-]+",S$
>>>
>>> ^(%(__prefix_line)s|\[\]\s*WARNING%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>>> )Ext\. s: "Rejecting unknown SIP connection from "$
>>>
>>> ignoreregex =
>>>
>>>
>>> # Author: Xavier Devlamynck / Daniel Black
>>> #
>>> # General log format - main/logger.c:ast_log
>>> # Address format - ast_sockaddr_stringify
>>> #
>>> # First regex: channels/chan_sip.c
>>> #
>>> # main/logger.c:ast_log_vsyslog - "in {functionname}:" only occurs in s
>>>
>>
>>
>> --
>> Technical Support
>> http://www.cellroute.net
>>
>>
>>
>> --
>> _
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Fail2ban

[Gokan Atmaca]

>>
>> I'm using the Fail2ban.  I configuration below. I want to try to
>> prevent the continuous password. Fail2ban password that does not
>> prevent this form. (Asterisk 1.8 / Elastix interface)
>>

hi

Asterisk version 1.8
Fail2ban version 0.8.14
  config: 
https://github.com/fail2ban/fail2ban/blob/master/config/filter.d/asterisk.conf

But it does not prevent.





On Sun, Sep 13, 2015 at 7:11 PM, Carlos Chavez <cur...@telecomabmex.com> wrote:
> On 2015-09-13 10:16, Gokan Atmaca wrote:
>>
>> Hello
>>
>> I'm using the Fail2ban.  I configuration below. I want to try to
>> prevent the continuous password. Fail2ban password that does not
>> prevent this form. (Asterisk 1.8 / Elastix interface)
>>
>> What could be the problem ?
>>
>> Asterisk log;
>> "Registration from '<sip:3...@sip.x.eu;transport=UDP>' failed for
>> 'x.x.x.x:32956' - Wrong password"
>>
>>
>> Fail2ban asterisk filter;
>>
>> # Fail2Ban filter for asterisk authentication failures
>> #
>>
>> [INCLUDES]
>>
>> # Read common prefixes. If any customizations available -- read them from
>>
>> # common.local
>> before = common.conf
>>
>>
>> [Definition]
>>
>> _daemon = asterisk
>>
>> __pid_re = (?:\[\d+\])
>>
>> # All Asterisk log messages begin like this:
>> log_prefix= (?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>> \S+:\d*( in \w+:)?
>>
>> failregex = ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Registration
>> from '[^']*' failed for '(:\d+)?' - (Wrong
>> password|Username/auth name mismatch|No m$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>> not found in context 'de$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed to authenticate as '[^']*'$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>> for peer '[^']*' \(from \)$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed MD5 authentication for '[^']*' \([^)]+\)$
>>   ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
>> '[^']*' \(:\d+\) to extension '\d+' rejected because extension
>> not found in context 'de$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed to authenticate as '[^']*'$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
>> for peer '[^']*' \(from \)$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
>> failed MD5 authentication for '[^']*' \([^)]+\)$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Failed to
>> authenticate (user|device) [^@]+@\S*$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>> (?:handle_request_subscribe: )?Sending fake auth rejection for
>> (device|user) \d*<sip:[^@]+@>;tag=$
>> ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
>>
>> SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="[\d-]+",S$
>>
>> ^(%(__prefix_line)s|\[\]\s*WARNING%(__pid_re)s:?(?:\[C-[\da-f]*\])?
>> )Ext\. s: "Rejecting unknown SIP connection from "$
>>
>> ignoreregex =
>>
>>
>> # Author: Xavier Devlamynck / Daniel Black
>> #
>> # General log format - main/logger.c:ast_log
>> # Address format - ast_sockaddr_stringify
>> #
>> # First regex: channels/chan_sip.c
>> #
>> # main/logger.c:ast_log_vsyslog - "in {functionname}:" only occurs in s
>
>
>  In the fail2ban website they have several versions of asterisk.conf
> depending on the version of Asterisk you are using.  If you have the latest
> fail2ban that one has the version for Asterisk 11.  Go there and download
> the correct version for your setup.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)9116-91161
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Fail2ban

[Gokan Atmaca]

Hello

I'm using the Fail2ban.  I configuration below. I want to try to
prevent the continuous password. Fail2ban password that does not
prevent this form. (Asterisk 1.8 / Elastix interface)

What could be the problem ?

Asterisk log;
"Registration from '' failed for
'x.x.x.x:32956' - Wrong password"


Fail2ban asterisk filter;

# Fail2Ban filter for asterisk authentication failures
#

[INCLUDES]

# Read common prefixes. If any customizations available -- read them from

# common.local
before = common.conf


[Definition]

_daemon = asterisk

__pid_re = (?:\[\d+\])

# All Asterisk log messages begin like this:
log_prefix= (?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[C-[\da-f]*\])?
\S+:\d*( in \w+:)?

failregex = ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Registration
from '[^']*' failed for '(:\d+)?' - (Wrong
password|Username/auth name mismatch|No m$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
'[^']*' \(:\d+\) to extension '\d+' rejected because extension
not found in context 'de$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
failed to authenticate as '[^']*'$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
for peer '[^']*' \(from \)$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
failed MD5 authentication for '[^']*' \([^)]+\)$
  ^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Call from
'[^']*' \(:\d+\) to extension '\d+' rejected because extension
not found in context 'de$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
failed to authenticate as '[^']*'$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s No registration
for peer '[^']*' \(from \)$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Host 
failed MD5 authentication for '[^']*' \([^)]+\)$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s Failed to
authenticate (user|device) [^@]+@\S*$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
(?:handle_request_subscribe: )?Sending fake auth rejection for
(device|user) \d*;tag=$
^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s
SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="[\d-]+",S$
^(%(__prefix_line)s|\[\]\s*WARNING%(__pid_re)s:?(?:\[C-[\da-f]*\])?
)Ext\. s: "Rejecting unknown SIP connection from "$

ignoreregex =


# Author: Xavier Devlamynck / Daniel Black
#
# General log format - main/logger.c:ast_log
# Address format - ast_sockaddr_stringify
#
# First regex: channels/chan_sip.c
#
# main/logger.c:ast_log_vsyslog - "in {functionname}:" only occurs in s

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Re: [asterisk-users] AstLinux 1.2.0 Released

[Gokan Atmaca]

On Thu, Oct 2, 2014 at 6:04 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
 The AstLinux Team has released 1.2.0. All current users are encouraged to 
 upgrade as this release addresses the bash ShellShock bug.

 New in 1.2.0:
 * New Linux Kernel 3.2.x
 * igb ethernet driver for Intel Atom C2000
 * Enable AES-NI support
 * New sip-user-agent firewall plugin
 * New versions of Asterisk 11 and 1.8
 * Bash ShellShock security fixes

 A full changelog can be viewed in the release pages:

 http://www.astlinux.org/release/120-asterisk-11121
 http://www.astlinux.org/release/120-asterisk-18300

 New AstLinux Documentation Topics:

 SMTP Local Aliases
 http://doc.astlinux.org/userdoc:tt_smtp_aliases

 Updated AstLinux Documentation Topics:

 Firewall Plugins - sip-user-agent
 http://doc.astlinux.org/userdoc:tt_firewall_plugins#sip-user-agent

 --The AstLinux Team

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Thanks for the info

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Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?

[Gokan Atmaca]

Someone reported me that from a PBX on which someone gained fraudulent
access, he could observe hundreds of calls to the same destination
number.

For curiosity's sake, I'm wondering why would this happen (dialing the
same number over and over) ?

Some special numbers generate here and there revenues for callees (and
not for callers).
Beside sharing interests with the callee that get those revenues, why
a hacker would like to dial the same numbers over and over ?
In other words, in this case, is looking at callee number a promising
path to find hackers ?

Is there a bot virus ? Do you IP address restrictions ?




On Wed, Oct 1, 2014 at 4:36 PM, Administrator TOOTAI ad...@tootai.net wrote:
 Le 01/10/2014 11:40, Olivier a écrit :

 Hi,


 Hi


 Someone reported me that from a PBX on which someone gained fraudulent
 access, he could observe hundreds of calls to the same destination
 number.

 For curiosity's sake, I'm wondering why would this happen (dialing the
 same number over and over) ?

 Some special numbers generate here and there revenues for callees (and
 not for callers).
 Beside sharing interests with the callee that get those revenues, why
 a hacker would like to dial the same numbers over and over ?


 callee is also the bad men. Go and buy an 899 number in France, hack PBXS
 and call your number :-)

 [...]

 --
 Daniel


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Re: [asterisk-users] AsteriskCDR

[Gokan Atmaca]

It seems you did not enable the mysql modules when you compiled Asterisk 11 
and thus the old modules from 1.8 are still in /usr/lib/modules/asterisk.  Go 
to the source code directory where you compiled and do a make menuconfig and 
make sure you enable cdr_mysql and app_mysql.  Then do another make  make 
install so the modules are copied.


Thanks

On Wed, Sep 24, 2014 at 7:38 PM, Carlos Chavez cur...@telecomabmex.com wrote:
 On 9/23/14, 10:38 PM, Gokan Atmaca wrote:

 Hello;

 I was using the 1.8 version of Asterisk. However, due to a problem I had to
 update. Update reporting system is broken when you have made. Current
 version 11.10. I installed the modules in the system for problems that are
 missing. I getting error as follows.


 ^[[A[Sep 24 03:16:50] WARNING[3624] loader.c: Module 'app_mysql.so' was not
 compiled with the same compile-time options as this version of Asterisk.
 [Sep 24 03:16:50] WARNING[3624] loader.c: Module 'app_mysql.so' will not be
 initialized as it may cause instability.
 [Sep 24 03:16:50] WARNING[3624] loader.c: Module 'app_mysql.so' could not be
 loaded.
 Asterisk to 11.12.0 app_mysql.so found. But it gives the following error.


 Then I found the necessary libraries for version 10.11. (app_mysql.so,
 res_config_mysql.so) Now the'm getting an error as follows.


 Error loading module 'app_mysql.so': libmysqlclient.so.16: cannot open
 shared object file: No such file or directory



 I want to do this without re-installing. Can you help ?

 It seems you did not enable the mysql modules when you compiled Asterisk
 11 and thus the old modules from 1.8 are still in /usr/lib/modules/asterisk.
 Go to the source code directory where you compiled and do a make
 menuconfig and make sure you enable cdr_mysql and app_mysql.  Then do
 another make  make install so the modules are copied.

 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52
 (55)9116-91161


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[asterisk-users] AsteriskCDR

[Gokan Atmaca]

Hello;

I was using the 1.8 version of Asterisk. However, due to a problem I had to
update. Update reporting system is broken when you have made. Current
version 11.10. I installed the modules in the system for problems that are
missing. I getting error as follows.


^[[A[Sep 24 03:16:50] WARNING[3624] loader.c:* Module 'app_mysql.so' was
not compiled with the same compile-time options as this version of
Asterisk.*
[Sep 24 03:16:50] WARNING[3624] loader.c: Module 'app_mysql.so' will not be
initialized as it may cause instability.
[Sep 24 03:16:50] WARNING[3624] loader.c: Module 'app_mysql.so' could not
be loaded.
Asterisk to 11.12.0 app_mysql.so found. But it gives the following error.


Then I found the necessary libraries for version 10.11. (app_mysql.so,
res_config_mysql.so) Now the'm getting an error as follows.


*Error loading module 'app_mysql.so': libmysqlclient.so.16*: cannot open
shared object file: No such file or directory



I want to do this without re-installing. Can you help ?
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