Re: [asterisk-users] how to stop hacking of my server

[Leandro Dardini]

2011/12/27 virendra bhati 

> Hi list someone is trying to hack my server . Is there any way by whcih I
> can stop hacking of my server except iptables ? I want to stop on the basis
> of sip.conf account only. bcoz I can't apply iptables rules on server it's
> remote server of server provider and we used it for making voip call for
> customers.
>
> for the time been i have close all sip accounts. but can't stop for more
> then 1 days. I need your help 
>
> *CLI log:- *
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed 

Re: [asterisk-users] execute command just after Dial()

[DHAVAL INDRODIYA]

You can also try special extension hangup and manage your scenario

On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind  wrote:

> Hi,
> Please see the Dial application documents from CLI, i.e "core show
> application dial". There is an option which will let you continue in the
> DIal-plan after the Dial command on hangup.
>
> Regards,
> Sammy.
>
> On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar wrote:
>
>>  Hello,
>>
>> I'm using AGI scripting with asterisk and need to execute certain
>> commands just after Dial(). But once dial command is executed, further
>> commands/instructions are ignored.
>>
>>
>> $agi->exec("Dial","SIP/100");
>> $dialstatus = $agi -> get_variable("DIALSTATUS");
>>
>> if($dialstatus[data]=="ANSWER")
>>
>> {
>>do something...
>> }
>>
>> thanks,
>> Kamlesh
>>
>> --
>> _
>> -- 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 --
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>
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Re: [asterisk-users] how to stop hacking of my server

[virendra bhati]

Can you give an example how to set these oprion ...


On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini  wrote:

>
>
> 2011/12/27 virendra bhati 
>
>> Hi list someone is trying to hack my server . Is there any way by whcih I
>> can stop hacking of my server except iptables ? I want to stop on the basis
>> of sip.conf account only. bcoz I can't apply iptables rules on server it's
>> remote server of server provider and we used it for making voip call for
>> customers.
>>
>> for the time been i have close all sip accounts. but can't stop for more
>> then 1 days. I need your help 
>>
>> *CLI log:- *
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
>> Registration from '"4411" ' failed for '
>> 62.141.54.169' - Wrong password
>> [Dec 26 21:21:21] NOT

Re: [asterisk-users] execute command just after Dial()

[DHAVAL INDRODIYA]

so you can try with options of dial application

 g: Proceed with dialplan execution at the next priority in the current
extension if the destination channel hangs up.


G([[context^]exten^]priority): If the call is answered, transfer
the calling party to the specified  and the called party to
the specified   plus one.
NOTE: You cannot use any additional action post answer options in
conjunction with this option.


On Tue, Dec 27, 2011 at 6:15 PM, Kamlesh Kumar wrote:

>  hangup extension works once the call is terminated but I want to know the
> status of call immediately after connected, cancelled, or rejected and so
> on.
>
> thanks,
> Kamlesh
>
>  --
> Date: Tue, 27 Dec 2011 16:59:35 +0530
> From: dhaval.it01...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] execute command just after Dial()
>
>
> You can also try special extension hangup and manage your scenario
>
> On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind  wrote:
>
> Hi,
> Please see the Dial application documents from CLI, i.e "core show
> application dial". There is an option which will let you continue in the
> DIal-plan after the Dial command on hangup.
>
> Regards,
> Sammy.
>
>  On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar 
> wrote:
>
>   Hello,
>
> I'm using AGI scripting with asterisk and need to execute certain commands
> just after Dial(). But once dial command is executed, further
> commands/instructions are ignored.
>
>
> $agi->exec("Dial","SIP/100");
> $dialstatus = $agi -> get_variable("DIALSTATUS");
>
> if($dialstatus[data]=="ANSWER")
>
> {
>do something...
> }
>
> thanks,
> Kamlesh
>
> --
> _
> -- 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:
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>
> --
> _
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[asterisk-users] read dtmf digits on connected calls

[Kamlesh Kumar]

Hello,
 
I need to capture the DTMF digits dialled by user on current connected calls 
and store them in variable. 
 
scenario:
Manual Call Transfer:
 
User A dialed to B
B answered the call and want to transfer the call to user C manually. User B 
dials *2 to get the ring tone again and then dial to 3. This works but I want 
to capture the digits *2 dialed by User B in some variable. Please suggest.
 
Regards,
Kamlesh
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Re: [asterisk-users] how to stop hacking of my server

[Leandro Dardini]

Yes, this is one of my entries:

[trunk1]
context=fromoutside
type=friend
deny=0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes

2011/12/27 virendra bhati 

> Can you give an example how to set these oprion ...
>
>
>
> On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini wrote:
>
>>
>>
>> 2011/12/27 virendra bhati 
>>
>>> Hi list someone is trying to hack my server . Is there any way by whcih
>>> I can stop hacking of my server except iptables ? I want to stop on the
>>> basis of sip.conf account only. bcoz I can't apply iptables rules on server
>>> it's remote server of server provider and we used it for making voip call
>>> for customers.
>>>
>>> for the time been i have close all sip accounts. but can't stop for more
>>> then 1 days. I need your help 
>>>
>>>
--
_
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Re: [asterisk-users] how to used SIPp for sip load testing

[Steve Murphy]

On Tue, Dec 27, 2011 at 6:33 AM, virendra bhati  wrote:

> Hi Sammy,
>
> I did the same and start calling. And it's working find but Now I want to
> the server max capacity with this script then what is the correct process..?
>

There is a nice tutorial on how you can do this in the asterisk source code:

./doc/chan_sip-perf-testing.txt

murf


>
>
> On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind  wrote:
>
>> Hi,
>> as the Logs say clearly you need to create an extension in default
>> context named service
>>
>> [default]
>> .
>> exten => service,1,NOOP(Incoming call from SIPp)
>> .
>>
>> Regards,
>> Sammy
>>
>>
>> On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati wrote:
>>
>>> Hi list,
>>>
>>> I have installed SIPp into my server. But not able to used it properly.
>>> how to configure with my server ? how to see logs on webpage ?
>>> how to start call testing 
>>>
>>> when i start SIPp then found verious hits on myserver.
>>>
>>> *CLI:- *
>>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>>   == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not found in
>>> context 'default'.
>>> haddock8-astrx*CLI>
>>>
>>>
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- 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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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Steve Murphy

ParseTree Corporation

57 Lane 17

Cody, WY 82414

✉  m...@parsetree.com

☎ 307-899-5535
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Re: [asterisk-users] how to stop hacking of my server

[Administrator TOOTAI]

Le 27/12/2011 16:04, Tim Nelson a écrit :

- Original Message -

On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati<  virbh...@gmail.com

wrote:



Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server except iptables ?

[...]

Odd nobody else mentioned it yet, so I'll do it...

Check out fail2ban. [...]


He said except iptables. fail2ban is iptables related ;-)

--
Daniel

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Re: [asterisk-users] how to used SIPp for sip load testing

[Sammy Govind]

Hi,
as the Logs say clearly you need to create an extension in default context
named service

[default]
.
exten => service,1,NOOP(Incoming call from SIPp)
.

Regards,
Sammy


On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati  wrote:

> Hi list,
>
> I have installed SIPp into my server. But not able to used it properly.
> how to configure with my server ? how to see logs on webpage ?
> how to start call testing 
>
> when i start SIPp then found verious hits on myserver.
>
> *CLI:- *
> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
>   == Using SIP RTP CoS mark 5
> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
> Call from '' to extension 'service' rejected because extension not found in
> context 'default'.
> haddock8-astrx*CLI>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
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Re: [asterisk-users] how to used SIPp for sip load testing

[virendra bhati]

Hi Sammy,

I did the same and start calling. And it's working find but Now I want to
the server max capacity with this script then what is the correct process..?

On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind  wrote:

> Hi,
> as the Logs say clearly you need to create an extension in default context
> named service
>
> [default]
> .
> exten => service,1,NOOP(Incoming call from SIPp)
> .
>
> Regards,
> Sammy
>
>
> On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati wrote:
>
>> Hi list,
>>
>> I have installed SIPp into my server. But not able to used it properly.
>> how to configure with my server ? how to see logs on webpage ?
>> how to start call testing 
>>
>> when i start SIPp then found verious hits on myserver.
>>
>> *CLI:- *
>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>>   == Using SIP RTP CoS mark 5
>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>> Call from '' to extension 'service' rejected because extension not found in
>> context 'default'.
>> haddock8-astrx*CLI>
>>
>>
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>


-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] how to stop hacking of my server

[Leandro Dardini]

With deny you'll "deny" all IP
with permit you'll "permit" only your IP.

Yes, it is mandatory to define both deny and permit.

Leandro

2011/12/27 virendra bhati 

> okay,
> So it is mandatory to define both permit and deny ?
> if I will update like
>
>
> [trunk1]
> context=fromoutside
> type=friend
> 
> permit=34.2.10.24
> qualify=yes
>
> So will it be fine or not ? Or it will get rest information from sip.conf
> general section ?
>
> On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini wrote:
>
>> Yes, this is one of my entries:
>>
>> [trunk1]
>> context=fromoutside
>> type=friend
>> deny=0.0.0.0/0.0.0.0
>> permit=34.2.10.24
>> qualify=yes
>>
>> 2011/12/27 virendra bhati 
>>
>>> Can you give an example how to set these oprion ...
>>>
>>>
>>>
>>> On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini wrote:
>>>


 2011/12/27 virendra bhati 

> Hi list someone is trying to hack my server . Is there any way by
> whcih I can stop hacking of my server except iptables ? I want to stop on
> the basis of sip.conf account only. bcoz I can't apply iptables rules on
> server it's remote server of server provider and we used it for making 
> voip
> call for customers.
>
> for the time been i have close all sip accounts. but can't stop for
> more then 1 days. I need your help 
>
>
>> --
>> _
>> -- 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
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- 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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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Joseph]

On 12/27/11 08:16, Ryan Wagoner wrote:

  On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling  wrote:

I'm getting various codec related warnings after upgrading to 1.8.  Did
I miss something in the UPGRADE file?  Does Asterisk no longer transcode
8-)?

WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native
formats 0x4 (ulaw)

And

WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel
SIP/interglobe-sip-01e6 setting write format to g722 from ulaw
native formats 0x4 (ulaw)

  When I upgraded to 1.8.7 I had a bunch of those warnings. I had backported
  the below patch which was included in a 1.8.8 release candidate. Since
  1.8.8 has been released I would just upgrade to that.

  https://issues.asterisk.org/jira/browse/ASTERISK-17541

  Ryan


Thanks folks, I've upgraded today to 1.8.8.0 and will test and report the 
result later on.

--
Joseph

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Re: [asterisk-users] how to stop hacking of my server

[Tim Nelson]

- Original Message -
> On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati < virbh...@gmail.com
> > wrote:
> 
> 
> 
> Hi list someone is trying to hack my server . Is there any way by
> whcih I can stop hacking of my server except iptables ? I want to stop
> on the basis of sip.conf account only. bcoz I can't apply iptables
> rules on server it's remote server of server provider and we used it
> for making voip call for customers.
> 

Odd nobody else mentioned it yet, so I'll do it...

Check out fail2ban. If you have peers or systems that you cannot restrict by IP 
and must leave relatively 'open', fail2ban will see the failed attempts, and 
after a configurable number of failures, will automatically add the offending 
IP to IPtables.

See here: 
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk

--Tim

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Re: [asterisk-users] how to stop hacking of my server

[Eric Wieling]

I suspect nobody responded because this topic has been discussed over and over 
again.  Search the mailing list archives.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator 
TOOTAI
Sent: Tuesday, December 27, 2011 11:34 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] how to stop hacking of my server

Le 27/12/2011 16:04, Tim Nelson a écrit :
> - Original Message -
>> On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati<  virbh...@gmail.com
>>> wrote:
>>
>>
>> Hi list someone is trying to hack my server . Is there any way by 
>> whcih I can stop hacking of my server except iptables ?
>>
>> [...]
> Odd nobody else mentioned it yet, so I'll do it...
>
> Check out fail2ban. [...]

He said except iptables. fail2ban is iptables related ;-)

--
Daniel

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Re: [asterisk-users] how to stop hacking of my server

[virendra bhati]

Thank you Leandro,

Now i am able to register with fix IP.


On Tue, Dec 27, 2011 at 3:10 PM, Leandro Dardini  wrote:

> With deny you'll "deny" all IP
> with permit you'll "permit" only your IP.
>
> Yes, it is mandatory to define both deny and permit.
>
> Leandro
>
>
> 2011/12/27 virendra bhati 
>
>> okay,
>> So it is mandatory to define both permit and deny ?
>> if I will update like
>>
>>
>> [trunk1]
>> context=fromoutside
>> type=friend
>> 
>> permit=34.2.10.24
>> qualify=yes
>>
>> So will it be fine or not ? Or it will get rest information from sip.conf
>> general section ?
>>
>> On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini wrote:
>>
>>> Yes, this is one of my entries:
>>>
>>> [trunk1]
>>> context=fromoutside
>>> type=friend
>>> deny=0.0.0.0/0.0.0.0
>>> permit=34.2.10.24
>>> qualify=yes
>>>
>>> 2011/12/27 virendra bhati 
>>>
 Can you give an example how to set these oprion ...



 On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini wrote:

>
>
> 2011/12/27 virendra bhati 
>
>> Hi list someone is trying to hack my server . Is there any way by
>> whcih I can stop hacking of my server except iptables ? I want to stop on
>> the basis of sip.conf account only. bcoz I can't apply iptables rules on
>> server it's remote server of server provider and we used it for making 
>> voip
>> call for customers.
>>
>> for the time been i have close all sip accounts. but can't stop for
>> more then 1 days. I need your help 
>>
>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>> --
>> _
>> -- 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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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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>



-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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[asterisk-users] how to used SIPp for sip load testing

[virendra bhati]

Hi list,

I have installed SIPp into my server. But not able to used it properly.
how to configure with my server ? how to see logs on webpage ?
how to start call testing 

when i start SIPp then found verious hits on myserver.

*CLI:- *
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
  == Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
haddock8-astrx*CLI>



-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Ryan Wagoner]

On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling  wrote:

> I'm getting various codec related warnings after upgrading to 1.8.  Did I
> miss something in the UPGRADE file?  Does Asterisk no longer transcode 8-)?
>
> WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
> DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native
> formats 0x4 (ulaw)
>
> And
>
> WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel
> SIP/interglobe-sip-01e6 setting write format to g722 from ulaw native
> formats 0x4 (ulaw)
>
>
When I upgraded to 1.8.7 I had a bunch of those warnings. I had backported
the below patch which was included in a 1.8.8 release candidate. Since
1.8.8 has been released I would just upgrade to that.

https://issues.asterisk.org/jira/browse/ASTERISK-17541

Ryan
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Re: [asterisk-users] how to stop hacking of my server

[Carlos Alvarez]

On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati  wrote:

> Hi list someone is trying to hack my server . Is there any way by whcih I
> can stop hacking of my server except iptables ? I want to stop on the basis
> of sip.conf account only. bcoz I can't apply iptables rules on server it's
> remote server of server provider and we used it for making voip call for
> customers.
>

Your iptables question has been answered, but I also wanted to comment on
SIP account naming.  Don't name them something obvious, particularly
numbers, since they will be attacked constantly.  Name your SIP peers
something that isn't going to be in the standard attack scripts.  We use
letters.numbers format to name them.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] execute command just after Dial()

[Kamlesh Kumar]

hangup extension works once the call is terminated but I want to know the 
status of call immediately after connected, cancelled, or rejected and so on.
 
thanks,
Kamlesh
 



Date: Tue, 27 Dec 2011 16:59:35 +0530
From: dhaval.it01...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] execute command just after Dial()

You can also try special extension hangup and manage your scenario


On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind  wrote:

Hi,
Please see the Dial application documents from CLI, i.e "core show application 
dial". There is an option which will let you continue in the DIal-plan after 
the Dial command on hangup.


Regards,
Sammy.




On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar  wrote:





Hello,
 
I'm using AGI scripting with asterisk and need to execute certain commands just 
after Dial(). But once dial command is executed, further commands/instructions 
are ignored.
 
 
$agi->exec("Dial","SIP/100");
$dialstatus = $agi -> get_variable("DIALSTATUS"); 
 
if($dialstatus[data]=="ANSWER")
   
{
   do something...
}
 
thanks,
Kamlesh

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Re: [asterisk-users] DIALSTATUS Values

[Kamlesh Kumar]

Hello,
 
After investing some time, I could come to know the reason for not getting the 
data value is that if I use system command with any of asterisk cli command as 
given below, data value is returned blank.
 
$output=system("/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d 
/ | grep '100' ")
 
Could you please suggest now how to rectify this?
 
Regards,
Kamlesh
 

> To: asterisk-users@lists.digium.com
> From: t...@softins.co.uk
> Date: Fri, 2 Dec 2011 12:27:19 +
> Subject: Re: [asterisk-users] DIALSTATUS Values
> 
> In article ,
> Kamlesh Kumar  wrote:
> > In addition to my reply:
> > 
> > I used to fetch the value using print_r function but that also tells that 
> > there is no value
> > in data section.
> > $dialstatus=$agi->get_variable(DIALSTATUS);
> > print_r($dialstatus);
> > 
> > AGI Rx << GET VARIABLE DIALSTATUS
> > AGI Tx >> 200 result=1 (CANCEL)
> > AGI Rx << Array
> > AGI Tx >> 510 Invalid or unknown command
> > [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
> > returned error: Broken pipe
> > AGI Rx << (
> > AGI Tx >> 510 Invalid or unknown command
> > [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
> > returned error: Broken pipe
> > AGI Rx << [code] => 200
> > AGI Tx >> 510 Invalid or unknown command
> > [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
> > returned error: Broken pipe
> > AGI Rx << [result] => 1
> > AGI Tx >> 510 Invalid or unknown command
> > [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
> > returned error: Broken pipe
> > AGI Rx << [data] =>
> 
> Well since the AGI return string does indeed contain the value, shown
> above as (CANCEL), that suggests there is definitely a bug in php-agi.
> It appears to be creating a ['data'] element, but not setting it.
> You will have to study the source code and work out how to fix it.
> I did a quick google for "php agi get variable" and found other reports
> of it not working properly, but I didn't see anyone offer a solution.
> It's only programming, so it shouldn't be hard to fix.
> 
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
> 
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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Eric Wieling]

We are running 1.8.8.0.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Tuesday, December 27, 2011 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec warnings after upgrade to 1.8

On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling  wrote:


I'm getting various codec related warnings after upgrading to 1.8.  Did 
I miss something in the UPGRADE file?  Does Asterisk no longer transcode 8-)?

WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel 
DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats 
0x4 (ulaw)

And

WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel 
SIP/interglobe-sip-01e6 setting write format to g722 from ulaw native 
formats 0x4 (ulaw)




When I upgraded to 1.8.7 I had a bunch of those warnings. I had backported the 
below patch which was included in a 1.8.8 release candidate. Since 1.8.8 has 
been released I would just upgrade to that.

https://issues.asterisk.org/jira/browse/ASTERISK-17541 

Ryan


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Re: [asterisk-users] how to stop hacking of my server

[virendra bhati]

okay,
So it is mandatory to define both permit and deny ?
if I will update like


[trunk1]
context=fromoutside
type=friend

permit=34.2.10.24
qualify=yes

So will it be fine or not ? Or it will get rest information from sip.conf
general section ?

On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini  wrote:

> Yes, this is one of my entries:
>
> [trunk1]
> context=fromoutside
> type=friend
> deny=0.0.0.0/0.0.0.0
> permit=34.2.10.24
> qualify=yes
>
> 2011/12/27 virendra bhati 
>
>> Can you give an example how to set these oprion ...
>>
>>
>>
>> On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini wrote:
>>
>>>
>>>
>>> 2011/12/27 virendra bhati 
>>>
 Hi list someone is trying to hack my server . Is there any way by whcih
 I can stop hacking of my server except iptables ? I want to stop on the
 basis of sip.conf account only. bcoz I can't apply iptables rules on server
 it's remote server of server provider and we used it for making voip call
 for customers.

 for the time been i have close all sip accounts. but can't stop for
 more then 1 days. I need your help 


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-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] how to stop hacking of my server

[Carlos Rojas]

Hello

I use fail2ban, and works fine,


Regards

On Tue, Dec 27, 2011 at 1:54 AM, virendra bhati  wrote:

> Hi list someone is trying to hack my server . Is there any way by whcih I
> can stop hacking of my server except iptables ? I want to stop on the basis
> of sip.conf account only. bcoz I can't apply iptables rules on server it's
> remote server of server provider and we used it for making voip call for
> customers.
>
> for the time been i have close all sip accounts. but can't stop for more
> then 1 days. I need your help 
>
> *CLI log:- *
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
> Registration from '"4411" ' failed for
> '62.141.54.169' - Wrong password
> [Dec 26 21:21:21] NOTICE[1770]: 

Re: [asterisk-users] odd "secret" problem

[sean darcy]

On 12/26/2011 10:05 PM, sean darcy wrote:

I've now set up tcp to connect for some home-office connections.

Home is 10.0.0, office is 1.8.8.0.

The home sip device is home-going-to-office, office device:
office-coming-from-home - home ip is 10.10.11.180

-- Called SIP/home-going-to-office/166
[Dec 26 18:42:31] NOTICE[4387]: chan_sip.c:20408 handle_response_invite:
Failed to authenticate on INVITE to '"HOME"
;tag=as3d6c8c47'

Nothing on the office cli.

home:
[home-going-to-office] ; places calls
type=peer ;; we only call out
transport=tcp
fromuser=office-coming-from-home
remotesecret=password
.

office:
[office-coming-from-home] ; receives calls
type=friend
transport=tcp
secret=password
..

What's odd is that if I remove secret from [office-coming-from-home] it
all works!!

sean



Have I missed something here? The "remotesecret" (I also tried just 
"secret" on the home side) should work, right?  Everything else must be 
ok since it works if I comment out "secret" on the office asterisk. 
Should I file a bug?


sean


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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Olivier]

2011/12/27, Eric Wieling :
> We are running 1.8.8.0.
>

Then the issue you're having differs from the one I had (which
appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8
respectively).

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Re: [asterisk-users] how to stop hacking of my server

[Tim Nelson]

- Original Message -
> Le 27/12/2011 16:04, Tim Nelson a écrit :
> > - Original Message -
> >> On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati<
> >> virbh...@gmail.com
> >>> wrote:
> >>
> >>
> >> Hi list someone is trying to hack my server . Is there any way by
> >> whcih I can stop hacking of my server except iptables ?
> >>
> >> [...]
> > Odd nobody else mentioned it yet, so I'll do it...
> >
> > Check out fail2ban. [...]
> 
> He said except iptables. fail2ban is iptables related ;-)
> 

Ahhh, yes, it would probably have helped if I read the message in it's 
entirety. :)

--Tim

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[asterisk-users] maximizing sound quality in 10.0

[Danny Nicholas]

Hi list,

I have a set of 300 or so WAV files I was combining and playing
using playback/background in 1.4.X.  Now that I have moved on to the 10.0
set, I understand that I can replace my 8 Khz mono files with virtually
unlimited Khz mono files (still no stereo, but a quantum leap forward).
I've played with this and get good throughputs using SLIN44 formats on SIP.
The 2 questions I have are:

1.   Is Slin44 the format I should be settling on or has someone found a
combination they find preferable?

2.   While the SIP connections sound good, I still have to "talk"
through OBI110 DADHI devices and other UUCM type connections - any pointers
for juicing up the sound there?

 

Thanks in Advance

Danny Nicholas

 

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[asterisk-users] Call going into s-extension

[Jonas Kellens]

Hello list,

any idea why this call goes to the extension 3292000101 :

/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: "32433885116" ;tag=74706
Via: SIP/2.0/UDP 
IP.IP.IP.IP:5060;rport;branch=z9hG4bKcc90522a02bb6591fd8a807e66
Via: SIP/2.0/UDP 
IP.IP.IP.IP:5060;branch=z9hG4bKd636e00c235e59dbb2d4a5eff83fdd96

Max-Forwards: 68
Content-Type: application/sdp
Contact: 
User-agent: Vox Callcontrol
To: 
Content-Length: 309
Record-Route: /


and this call goes to the s-extension :

/<--- SIP read from UDP:IP.IP.IP.IP:5060 --->
INVITE sip:s...@ip.ip.ip.ip:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bK918e.b1b09812.0
Via: SIP/2.0/UDP 
217.111.202.80:5060;received=217.111.202.80;branch=z9hG4bK5cb88830;rport=5060

Max-Forwards: 69
From: "32433885116" ;tag=as6aa259f4
To: 
Contact: 
Call-ID: 092fd61b1e3aad5e670dfa94578db96d@217.111.202.80:5060
CSeq: 102 INVITE
User-Agent: TLDOSIPS
Date: Tue, 27 Dec 2011 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296/


What is the difference ??


Kind regards,
Jonas.
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Re: [asterisk-users] Call going into s-extension

[Kevin P. Fleming]

On 12/27/2011 01:43 PM, Jonas Kellens wrote:

Hello list,

any idea why this call goes to the extension 3292000101 :

/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: "32433885116" ;tag=74706
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;rport;branch=z9hG4bKcc90522a02bb6591fd8a807e66
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;branch=z9hG4bKd636e00c235e59dbb2d4a5eff83fdd96
Max-Forwards: 68
Content-Type: application/sdp
Contact: 
User-agent: Vox Callcontrol
To: 
Content-Length: 309
Record-Route: /


and this call goes to the s-extension :

/<--- SIP read from UDP:IP.IP.IP.IP:5060 --->
INVITE sip:s...@ip.ip.ip.ip:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bK918e.b1b09812.0
Via: SIP/2.0/UDP
217.111.202.80:5060;received=217.111.202.80;branch=z9hG4bK5cb88830;rport=5060
Max-Forwards: 69
From: "32433885116" ;tag=as6aa259f4
To: 
Contact: 
Call-ID: 092fd61b1e3aad5e670dfa94578db96d@217.111.202.80:5060
CSeq: 102 INVITE
User-Agent: TLDOSIPS
Date: Tue, 27 Dec 2011 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296/


What is the difference ??


Did you look at the Request-URIs specified in the INVITE lines at the 
beginning of the SIP messages? One specifies 's' as the target, the 
other specifies '3292000101'.


--
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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

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Re: [asterisk-users] Call going into s-extension

[Jonas Kellens]

On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:

On 12/27/2011 01:43 PM, Jonas Kellens wrote:

Hello list,

any idea why this call goes to the extension 3292000101 :

/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: "32433885116" ;tag=74706
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;rport;branch=z9hG4bKcc90522a02bb6591fd8a807e66
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;branch=z9hG4bKd636e00c235e59dbb2d4a5eff83fdd96
Max-Forwards: 68
Content-Type: application/sdp
Contact: 
User-agent: Vox Callcontrol
To: 
Content-Length: 309
Record-Route: /


and this call goes to the s-extension :

/<--- SIP read from UDP:IP.IP.IP.IP:5060 --->
INVITE sip:s...@ip.ip.ip.ip:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bK918e.b1b09812.0
Via: SIP/2.0/UDP
217.111.202.80:5060;received=217.111.202.80;branch=z9hG4bK5cb88830;rport=5060 


Max-Forwards: 69
From: "32433885116" ;tag=as6aa259f4
To: 
Contact: 
Call-ID: 092fd61b1e3aad5e670dfa94578db96d@217.111.202.80:5060
CSeq: 102 INVITE
User-Agent: TLDOSIPS
Date: Tue, 27 Dec 2011 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296/


What is the difference ??


Did you look at the Request-URIs specified in the INVITE lines at the 
beginning of the SIP messages? One specifies 's' as the target, the 
other specifies '3292000101'.


Yes I saw these... why is there a difference ?


Kind regards,
Jonas.


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Re: [asterisk-users] Call going into s-extension

[Kevin P. Fleming]

On 12/27/2011 01:51 PM, Jonas Kellens wrote:

On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:

On 12/27/2011 01:43 PM, Jonas Kellens wrote:

Hello list,

any idea why this call goes to the extension 3292000101 :

/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: "32433885116" ;tag=74706
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;rport;branch=z9hG4bKcc90522a02bb6591fd8a807e66
Via: SIP/2.0/UDP
IP.IP.IP.IP:5060;branch=z9hG4bKd636e00c235e59dbb2d4a5eff83fdd96
Max-Forwards: 68
Content-Type: application/sdp
Contact: 
User-agent: Vox Callcontrol
To: 
Content-Length: 309
Record-Route: /


and this call goes to the s-extension :

/<--- SIP read from UDP:IP.IP.IP.IP:5060 --->
INVITE sip:s...@ip.ip.ip.ip:5060 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP IP.IP.IP.IP;branch=z9hG4bK918e.b1b09812.0
Via: SIP/2.0/UDP
217.111.202.80:5060;received=217.111.202.80;branch=z9hG4bK5cb88830;rport=5060

Max-Forwards: 69
From: "32433885116" ;tag=as6aa259f4
To: 
Contact: 
Call-ID: 092fd61b1e3aad5e670dfa94578db96d@217.111.202.80:5060
CSeq: 102 INVITE
User-Agent: TLDOSIPS
Date: Tue, 27 Dec 2011 19:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296/


What is the difference ??


Did you look at the Request-URIs specified in the INVITE lines at the
beginning of the SIP messages? One specifies 's' as the target, the
other specifies '3292000101'.


Yes I saw these... why is there a difference ?


You'd have to ask the devices that generated them. We aren't all-knowing 
and all-seeing. They are also different in quite a few other ways, 
including User-Agent, supported request methods, SIP extensions and others.


--
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

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Re: [asterisk-users] Locally bridging channels when using SRTP?

[Kevin P. Fleming]

On 12/01/2011 03:48 PM, Jan Blom wrote:

Hello,

I’m trying to setup an Asterisk (version 1.8.8) to do SRTP termination
and then send the call on to other servers, unencrypted. All the basics
work fine.

I want the Asterisk to do as little as possible with the RTP packets and
no transcoding. We always make sure to force same codec on incoming and
outgoing call leg.

When not using SRTP, Asterisk does P2P bridging of the RTP packets. That
is, simply copying the packets, which is the expected result. But when
we send in SRTP media, Asterisk starts decode/encode voice data instead
of just do P2P bridging.

I also notice Asterisk doesn’t say “Locally bridging channels” in the
latter case, which might be the clue that we’re not doing P2P bridging.

Why can we not use P2P bridging when doing SRTP->RTP media conversion?
Is there anything we can change in the source code to force packet
bridging in this case?


The payloads of the RTP packets on the either side of the bridge are not 
compatible with each other in this scenario. It's possible that what are 
asking for could be done, but the SRTP code in Asterisk isn't currently 
structured in a way that should allow for it.


However, the CPU cost difference between P2P bridging and core bridging 
is not that great, generally. There is no 'decode/encode' of the voice 
data unless that is required; core bridging just means that the incoming 
RTP packets are broken down and their payloads are copied into internal 
frames before turned back into RTP packets on the way out. There is some 
cost associated with this, but unless you are running a system that is 
right on the edge of falling over due to channel load, it should be 
tolerable.


--
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

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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

[Danny Nicholas]

Change requirecalltoken from auto to no.  1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, December 25, 2011 4:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

After upgrading one of my server to asterisk 1.8.7.2  (the older is running
1.4.39)

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.

On asterisk-1.8.7 I get:
  WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by
192.168.141.1: Unable to negotiate codec


I'm using ulaw / alaw code; why don't they communicate?

iax.conf (1.4.39)
[home_server]
disallow=all
allow=ulaw
allow=alaw

iax.conf (1.8.7)
[clinic_server]
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=auto

--
Joseph

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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

[John Novack -W7P]

Asterisk 1.4 after, or beginning with, 1.4.26 DOES know about call 
tokens, so this must be upset about something else


John Novack


Danny Nicholas wrote:

Change requirecalltoken from auto to no.  1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, December 25, 2011 4:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

After upgrading one of my server to asterisk 1.8.7.2  (the older is running
1.4.39)

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.

On asterisk-1.8.7 I get:
   WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by
192.168.141.1: Unable to negotiate codec


I'm using ulaw / alaw code; why don't they communicate?

iax.conf (1.4.39)
[home_server]
disallow=all
allow=ulaw
allow=alaw

iax.conf (1.8.7)
[clinic_server]
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=auto

--
Joseph

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Re: [asterisk-users] maximizing sound quality in 10.0

[Paulo Santos]

On Ter, 2011-12-27 at 13:34 -0600, Danny Nicholas wrote:
> Hi list,
> 
> I have a set of 300 or so WAV files I was combining and
> playing using playback/background in 1.4.X.  Now that I have moved on
> to the 10.0 set, I understand that I can replace my 8 Khz mono files
> with virtually unlimited Khz mono files (still no stereo, but a
> quantum leap forward).  I’ve played with this and get good throughputs
> using SLIN44 formats on SIP.   The 2 questions I have are:
> 
> 1.   Is Slin44 the format I should be settling on or has someone
> found a combination they find preferable?
> 
> 2.   While the SIP connections sound good, I still have to “talk”
> through OBI110 DADHI devices and other UUCM type connections – any
> pointers for juicing up the sound there?

I'm not sure, but unless you're strictly talking through SIP _and_ using
devices that support more than 8KHz, you won't take advantage of more
than that.

I believe that ISDN (BRI or PRI) and analogue lines as well as most
phones use 8KHz.

But, like I said, I'm not sure. Maybe someone can confirm that? I got
curious myself.


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[asterisk-users] cdr call time

[Vinod Dharashive]

Hi team,

On event of no answer in CDR the starttime and endtime of call remains the same.

Is there any way how can actually track call originate time and call end time.

Thanks
Vinod dharashive.


Sent from BlackBerry® on Airtel
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[asterisk-users] sendvoicemail=yes not quite working SOLVED

[M Maki]

Was missing the option searchcontexts=yes in voicemail.conf

Mike

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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Joseph]

On 12/27/11 18:23, Olivier wrote:

2011/12/27, Eric Wieling :

We are running 1.8.8.0.



Then the issue you're having differs from the one I had (which
appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8
respectively).



Upgrading to 1.8.8 DID NOT HELP
I'm getting the same error message:
Call rejected by ... Unable to negotiate codec

  == Using UDPTL CoS mark 5
  == Using SIP RTP CoS mark 5
-- Executing [4@internal:1] Dial("SIP/11-", 
"IAX2/home_server:@192.168.141.1/4,30,rw") in new stack
-- Called IAX2/home_server:@192.168.141.1/4
[Dec 27 19:10:42] WARNING[15968]: chan_iax2.c:10672 socket_process: Call 
rejected by 192.168.141.1: Unable to negotiate codec
-- Hungup 'IAX2/192.168.141.1:4569-24'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [4@internal:2] Hangup("SIP/11-", "") in new stack
  == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-'


--
Joseph

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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

[Joseph]

No, it makes no difference, on the other end is asterisk 1.4.39

and 1.8.8 is still giving me:

 Executing [4@internal:1] Dial("SIP/11-0003", 
"IAX2/home_server:@192.168.141.1/4,30,rw") in new stack
-- Called IAX2/home_server:@192.168.141.1/4
[Dec 27 20:00:16] WARNING[16398]: chan_iax2.c:10672 socket_process: Call 
rejected by 192.168.141.1: Unable to negotiate codec
-- Hungup 'IAX2/192.168.141.1:4569-5678'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [4@internal:2] Hangup("SIP/11-0003", "") in new stack
  == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0003'

--
Joseph


On 12/27/11 15:56, Danny Nicholas wrote:

Change requirecalltoken from auto to no.  1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, December 25, 2011 4:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

After upgrading one of my server to asterisk 1.8.7.2  (the older is running
1.4.39)

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.

On asterisk-1.8.7 I get:
 WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by
192.168.141.1: Unable to negotiate codec


I'm using ulaw / alaw code; why don't they communicate?

iax.conf (1.4.39)
[home_server]
disallow=all
allow=ulaw
allow=alaw

iax.conf (1.8.7)
[clinic_server]
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=auto

--
Joseph

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--
Joseph

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Re: [asterisk-users] how to stop hacking of my server

[virendra bhati]

Yes Eric,

I read the archive and found that all guys was saying another open sources
project for protection on server like fail2ban. But I want security at
configuration level only. As *Leandro* suggest permit and deny option of
Sip.conf and *Carlos* suggest the naming process. like that someone suggest
that naming should be the SIP phone MAC address. All these are the best for
starting security at configuration level.

thanks all who posted in this thread.
I will used and try Fail2ban but on another server.

On Tue, Dec 27, 2011 at 11:19 PM, Tim Nelson  wrote:

> - Original Message -
> > Le 27/12/2011 16:04, Tim Nelson a écrit :
> > > - Original Message -
> > >> On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati<
> > >> virbh...@gmail.com
> > >>> wrote:
> > >>
> > >>
> > >> Hi list someone is trying to hack my server . Is there any way by
> > >> whcih I can stop hacking of my server except iptables ?
> > >>
> > >> [...]
> > > Odd nobody else mentioned it yet, so I'll do it...
> > >
> > > Check out fail2ban. [...]
> >
> > He said except iptables. fail2ban is iptables related ;-)
> >
>
> Ahhh, yes, it would probably have helped if I read the message in it's
> entirety. :)
>
> --Tim
>
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-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] cdr call time

[Zohair Raza]

may this helps,

In cdr.conf, set endbeforehexten=yes

Regards,
Zohair Raza



On Wed, Dec 28, 2011 at 4:46 AM, Vinod Dharashive wrote:

> Hi team,
>
> On event of no answer in CDR the starttime and endtime of call remains the
> same.
>
> Is there any way how can actually track call originate time and call end
> time.
>
> Thanks
> Vinod dharashive.
>
>
> Sent from BlackBerry® on Airtel
> --
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