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Calling from extension that I don’t create

Dear Moderator,

I need an assistance for below problem:
1. there are time incoming call from extension 100 that I don’t create, how
to know where is the origin of this extension?
2. when I pick up the call, there is no sound at all, this is very iritating
me due sometime the call is made during midnight until dawn, how to handle
this?

Thank you for your help and assistance.

Best regards,
Eko Kukuh Wibowo
+62818325598

http://www.spidermetrix.com/sm120.php?refspider=flickvideo

Subscribe Problem – Zombie Channel

On Tuesday, 7 February 2012, Örn Arnarson wrote:
> Hi Brian,
>
> Did you ever figure out what’s causing this, and how to deal with it?
>
> I’m seeing the same behavior with call-pickups (it’s rare, but it’s
> happened a few times) on Asterisk 1.6.1.11
>
> Did you figure out a way to get rid of the channel without restarting?
>
> Regards,
>

Hi Orn

I didn’t find a way except a restart once active calls drop to zero.

Regards,
Brian

>
> On Wed, Jul 28, 2010 at 9:45 PM, dotnetdub wrote:
>>
>>
>> On 28 July 2010 21:42, Stefan Schmidt wrote:
>>>
>>> dotnetdub schrieb:
>>> > Hi List,
>>>
>>> > core show channels
>>> > Channel Location State Application(Data)
>>> >
>>> > SIP/102–08e1 *8@from-inside Down (None)
>>> > SIP/102–08d6 *8@from-inside Ring (None)
>>> > SIP/102–08d7 *8@from-inside Ring (None)
>>> > 3 active channels
>>> > 0 active calls
>>> >
>>> > The only way to free them up is to force a restart.
>>> >
>>> > restart now
>>> >
>>> > Any clues on how I can debug this and try to sort it or even if anyone
>>> > has come across this.
>>> >
>>> > Many thanks in advance.
>>> >
>>> > Brian
>>> >
>>> >
>>> hello,
>>>
>>> you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think
>>> you will see some locks when this happens with core show locks.
>>> how do you make the pickup? do you use an extension *8 for this, or just
>>> the feature for pickup in features conf?
>>>
>>> best regards
>>>
>>> steve
>>>
>>>
>>
>> Hi Steve,
>>
>> Thanks for the reply. We have:
>>
>> pickupexten = *8 ; Configure the pickup extension.
Default
>> is *8
>>
>> in features.conf.
>>
>> I will recompile on one of the sites this happens on. It’s really odd,
can
>> go for weeks without this happening and then a customer will report to me
>> that their extension is showing in use and I will login and there can be
two
>> or three of these locks. On one site it actually makes asterisk
impossible
>> to stop and I need to kill -9
>>
>> We have stuck with version 1.4.22 as it has been so solid for us, no
dumps
>> or deadlocks etc. We have tried to move to 1.4.25 and 1.4.29 but would
>> experience random weirdness that we just don’t get with this version.
>>
>> When recompiled with this flag and if indeed it does show locks, what
would
>> be the next step?
>>
>> Thanks for your help.
>> Brian
>>
>>
>> –
>> _____________________________________________________________________
>> — 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
>
> –
> _____________________________________________________________________
> — 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

dial a queue

No :(

2012/2/6, Danny Nicholas :
> Queue(8888)?
>
> —–Original Message—–
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alejandro
> Cabrera Obed
> Sent: Monday, February 06, 2012 1:40 PM
> To: Asterisk Users Mailing List – Non-Commercial Discussion
> Subject: Re: [asterisk-users] Custom extension: dial a queue
>
> No, Local/queue/8888 don’t work at all :(
>
> 2012/2/6, Danny Nicholas
:
>> Local/queue/8888?
>>
>> —–Original Message—–
>> From: asterisk-users-bounces@lists.digium.com
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
>> Alejandro Cabrera Obed
>> Sent: Monday, February 06, 2012 1:19 PM
>> To: Asterisk Users Mailing List – Non-Commercial Discussion
>> Subject: [asterisk-users] Custom extension: dial a queue
>>
>> Dear, I need to create a custom device extension in order to dial a
>> local queue.
>>
>> Suppose my queue number is 8888, how can fill the Dial field from the
>> custom extension ???
>>
>> Because if I put just 8888 or Local/8888, I don’t succeed.
>>
>> Thanks a lot
>>
>> –
>> _____________________________________________________________________
>> — 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
>>
>
>
> –
> Alejandro Cabrera Obed
> aco1967@gmail.com
> www.alejandrocabrera.com.ar
>
> –
> _____________________________________________________________________
> — 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
>

read digits during recording / DTMF in conference?

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they’d be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they’re too slow and recording has
started, they’ve missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It’s a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked – Read() managed to read the
DTMF digits and assign them to a variable.

However, when the ‘recording’ call is also in the conference, the ‘read’
call can no longer recognise the DTMF digits. To test, I made the ‘read’
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn’t notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

Sip Registration Hijacking

I have a honey pot box with extensions that are not just numbers ie )

100-MySipUserName

And the passwords are from an openssl generated password ie)

Gq5VNIjDFWIQoUT6

However, this one extension keeps getting hacked and showing up on a different IP address.

It is also register on an AudioCodes MP-118.

I wanted to know if anyone else ran into this and if it’s a vulnerability on the MP-118 or with Asterisk.

Thanks,

-E