That's a good point. I just checked and the vanity DID I really want
is not available in any area code. So either I'll come up with
something else or just stick with my acceptable but not optimal vanity
toll free. Thanks!
--
Tracy Reed
http://tracyreed.org
Tracy Reed wrote:
> What's the best way to get a good vanity toll free number? Or is it
> pointless? All of the good ones seem to be taken by phone sex
> operators or something. It's similar to the top level domain name
> thing: They are cheap to own so people squat them and use up all the
> namespace.
>
> I've googled and found various places which claim they can get me
> vanity toll free numbers and they all feel rather sleezy. Most seem to
> lead you on to think the number you want is available and then end up
> pitching you something like 877-161-216MYCOMPANY.
>
> I've done business with one toll-free broker and have mixed feelings
> about the result.
>
>
>
> ------------------------------------------------------------------------
>
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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
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I've googled and found various places which claim they can get me
vanity toll free numbers and they all feel rather sleezy. Most seem to
lead you on to think the number you want is available and then end up
pitching you something like 877-161-216MYCOMPANY.
I've done business with one toll-free broker and have mixed feelings
about the result.
--
Tracy Reed
http://copilotco.com
Darren Wiebe
darren@aleph-com.net
John Todd wrote:
> On Feb 27, 2009, at 1:04 PM, voip-asterisk@maximumcrm.com wrote:
>
>
>>>> I'd suggest to everyone to ban that IP, it's been scanning our
>>>> networks
>>>> from time to time, in a sequential manner by IP.
>>>>
>>> I've had really good luck with this:
>>>
>>> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
>>>
>>> Basically, it automatically blackhols via IPtables any host that
>>> fails a
>>> certain number of registration attempts in a given period.
>>>
>> Yeah we're actually rolling it out on all of our production servers,
>> it's
>> a great application to run.
>>
>> I'm working on some scripts to propagate the bans to the firewall so
>> that
>> all of the servers get protected as soon as possible.
>>
>>
>>> [default]
>>> ; Send any unauthenticated calls to the local FBI office
>>> context=local-fbi-office
>>>
>>> I've got a honeypot server that pretty much accepts any calls that
>>> come
>>> through, and plays a "Thank you for calling the Telecommunications
>>> Fraud
>>> hotline. Please stay online for the next available representative."
>>> If they
>>> stay online for more than 20 seconds, it connects them to an agent
>>> at the
>>> FBI that we have been working with.
>>>
>>> I've been meaning to add some code in that pulls out the
>>> originating IP
>>> address of the call and tells it to the agent when we call. :)
>>>
>> That would be great to have!
>>
>
>
>
> This sounds very much like the framework I discussed at the last
> astridevcon in September. I've had no time to work on it, but it
> sounds like you're already making progress.
>
> http://astridevcon.pbwiki.com/Network-Security-Framework
>
> Would you be interested in making your work more integral to Asterisk,
> so that it can be a generic security policy model for all channel
> methods, starting with SIP? Or is the scrape-from-logfile method
> sufficient for your needs?
>
> JT
>
>
> ---
> John Todd email:jtodd@digium.com
> Digium, Inc. | Asterisk Open Source Community Director
> 445 Jan Davis Drive NW - Huntsville AL 35806 - USA
> direct: +1-256-428-6083 http://www.digium.com/
>
>
>
>
> _______________________________________________
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>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
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We run a small number of Asterisk servers in a data centre and are noticing glitches in the audio. These glitches are consistent (i.e. the same type of problem) and happen frequently at random intervals. The audio problem takes the form of a dropout in the audio that is audible to the customer. We need some help in diagnosing the cause of these audio issues which we believe are network related (a recording of the milliwatt application sounds fine). These servers all run Asterisk 1.2 and use SIP only (no ISDN, IAX etc).
If you are able to provide some consulting skills in this area please contact me off list and advise relevant experience, references, rate, availability and approach.
Thanks
>>> I'd suggest to everyone to ban that IP, it's been scanning our
>>> networks
>>> from time to time, in a sequential manner by IP.
>>
>> I've had really good luck with this:
>>
>> http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
>>
>> Basically, it automatically blackhols via IPtables any host that
>> fails a
>> certain number of registration attempts in a given period.
>
> Yeah we're actually rolling it out on all of our production servers,
> it's
> a great application to run.
>
> I'm working on some scripts to propagate the bans to the firewall so
> that
> all of the servers get protected as soon as possible.
>
>> [default]
>> ; Send any unauthenticated calls to the local FBI office
>> context=local-fbi-office
>>
>> I've got a honeypot server that pretty much accepts any calls that
>> come
>> through, and plays a "Thank you for calling the Telecommunications
>> Fraud
>> hotline. Please stay online for the next available representative."
>> If they
>> stay online for more than 20 seconds, it connects them to an agent
>> at the
>> FBI that we have been working with.
>>
>> I've been meaning to add some code in that pulls out the
>> originating IP
>> address of the call and tells it to the agent when we call. :)
>
> That would be great to have!
This sounds very much like the framework I discussed at the last
astridevcon in September. I've had no time to work on it, but it
sounds like you're already making progress.
http://astridevcon.pbwiki.com/Network-Security-Framework
Would you be interested in making your work more integral to Asterisk,
so that it can be a generic security policy model for all channel
methods, starting with SIP? Or is the scrape-from-logfile method
sufficient for your needs?
JT
---
John Todd email:jtodd@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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