Re: [asterisk-users] VOIP & PBX replacement suggestions?

[Cary Fitch]

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel
Seagraves
Sent: Wednesday, June 06, 2012 2:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VOIP & PBX replacement suggestions?

The boss wants to move from landline service to VOIP service as a
cost-cutting measure. We have one voice line and one fax line. The telco is
billing over $100 a month for the two. We're using Hylafax for faxing and a
PBX for the voice line.

Our existing PBX is an Intertel Axxess box with the old v5 processor. The
management and voicemail computer died years ago (PSU burned up). I'm
worried that it's going to die before too much longer. We have the IPRC and
several IP Phone+ devices. It's my understanding that the IP Phone+ speaks
only a proprietary Intertel protocol and can never be used with any
non-Intertel equipment. I would like to dump the entire Intertel box and
move to Asterisk instead, but my budget for this project is exactly $0. I
can't afford to buy new devices.

The boss is leaning toward getting digital voice service from the local
cable monopoly. They want to charge us $30 a month per line to start, and we
will have to sign a 3 year contract. The monopoly in question has a
reputation for very poor service, but they are a monopoly so my boss sees
them as the only alternative. My worry is that if we sign that contract, we
are trapped with both the intertel and the cable monopoly, and if I exceed
the capacity of the intertel (or it just dies) I am SOL.

My questions then are as follows:

1) Is there a way I don't know about to get Asterisk to talk to either the
IPRC or the IP Phone+ directly in such a way that gets calls from one to the
other?
2) Are there any reputable VOIP providers that provide business service at a
rate less than $30 per line per month? The boss is adamant that we need
unlimited minutes.

===

Where do you get your IP connection?  The cable monopoly?

There are several companies you can get service from.
One is Teliax.com

Cary


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Re: [asterisk-users] OT - Incoming fax cuts ADSL line

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp
Sent: Wednesday, May 16, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT - Incoming fax cuts ADSL line

On 5/16/2012 12:07 PM, Tim Nelson wrote:
> - Original Message -
>> Hi,
>>
>> I'm facing a strange situation.
>> Though it's not directly related to Asterisk, I do think it is
>> interesting to this mailing list.
>>
>>
>> The setup is a single line which is split between an ADSL
>> modem/routeur and a fax machine (Asterisk was removed from the
>> equation).
>>
>> Any time the fax machine rings (incoming fax), the ADSL service is
>> troubled to the VPN users are disconnected.
>> It can be reproduced at will.
>>
>> I've changed the ADSL filter twice (a different unit, then a
>> different
>> model) without any visible change.
>> What could explain this ?
>>
>
> I've experienced this quite a few times, and after working with a local
telco, it has become policy to not place ADSL on lines where fax is going to
be used. I'm unsure of the exact technical reasons behind this other than
'the fax signals/frequencies interfere with the ADSL signalling/frequencies
used on the circuit'. It sounds like you might want to separate your
fax/ADSL lines.
>
> --Tim

You might also be able to limit the Fax machines maximum transmission 
rate so the modem's transmission spectrum doesn't inch up into where the 
ADSL service is.


ADSL is transmitted at a relatively low frequency using phase modulated
carriers to achieve the bandwidth. It could be about 32 different
phase/level locations on 360 degree/level "pie chart" or vector scope.

The actual frequencies of the carrier are moderately low, maybe 100 to 200
kcps.

Voice is low density.  Faxes and modems are high density and "loud".  They
can splatter or have harmonics that can confuse the local DSL demodulator.

As others have said, the best thing to try is the best filters you can get
between the phone line and the DSL demod, and maybe two filters in series.
If that doesn't work, put the fax on a different line than the DSL, which
could cost you money.  Paying for better filters or two of them is less
expensive than separate lines.

Or move the DSL to an alternate existing voice only line, since you probably
don't want to change the fax number

Contents of this message were dredged from foggy memory.

Cary Fitch





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Re: [asterisk-users] How to play audio file in background indialplan?

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav
Panych
Sent: Friday, February 03, 2012 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to play audio file in background
indialplan?

2012/2/3 Danny Nicholas :
> In my PERL AGI I use
> - print STDOUT "SET MUSIC ON HOLD DEFAULT\n";
> - print STDOUT "SET MUSIC ON HOLD OFF\n";
>
> Ignore the "-" - stupid outlook needs them.

There is one problem: I have not any MOH class, and cannot pre-create
it. File I will play does not exists until incoming call arrives, I
even don't know its exact name.


I am just an onlooker but could you explain that.  What are you wanting
to play? Where will it come from?  I am not sure how anyone can answer your
question given you don't have a file, can't pre-create it etc.  Are you
going to use the calling number as part of the file, or what?  What is the
caller going to hear?

CF




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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Cary Fitch]

We have run some tests on the Xorcom equipment, mostly the PRI port cards,
running up to 16 ports in a chassis.  They work.  I see no problem in Xorcom
as FXO ports.

 

We will be installing a lot of them as PRI ports soon..

 

CF

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco
Signorini
Sent: Wednesday, August 31, 2011 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] USB or Ethernet based FXO device ?

 

Hi.

I was following this thread. We normally use Patton SmartNode SN4112 series
to interface to FXO ports. But I'm looking for something different for a
future setup.
Xorcom USB channel banks seems something quite interesting. Is there anyone
that could/would share experiences using that? 
We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?

Thank you and regards,
Marco Signorini.

  -- 

http://www.ingegnitech.com/images/logo.gif

INGEGNI Tech S.r.l.
site   http://www.ingegnitech.com
mail   i...@ingegnitech.com

  _  

 



Gilles wrote: 

On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
   wrote:
  

  Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.


 
Thanks for the tip. It looks like the smallest option is 8 FXO ports:
 
www.xorcom.com/telephony-interfaces/astribank-models.html
 
 
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Re: [asterisk-users] Dragging the dialup customers along, possible?

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Monday, August 29, 2011 3:18 PM
To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Dragging the dialup customers along, possible?

>From what you are asking it appears that you are trying to run similar
to a fax (modulation and demodulation) over VoIP.
Try again, the fact that you succeeded twice was pure luck, and as far
as I understand that didn't even work out.
Switch back to TDM. Your dial up modems want that magic thing called
timing and no jitter that only TDM will give you.

===

This is more of a whimsical statement than a scientific one, but I would
think in today's world, there would be a real small box that would take in
IP and put out TDM with good timing with a moderate buffering window.
Obviously, the IP has to actually get to the box in a timely fashion, like
"today" , but a TDM circuit has to be "up" also.  

A box that would take in IP data..., look for valid "ascii", and otherwise
put out TDM modem tones with no data content for "1 second" and then pick up
the data as it catches up.

Better a laggy modem connection than no data at all.

CF


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Re: [asterisk-users] Securing Asterisk

[Cary Fitch]

CDR wrote:
> 
> The point is that a minor change in the code would have a dramatic
> effect on security, and carry a lower impact on CPU that using
> Iptables. The simplicity of the change cannot understated.


You're in luck.  Since Asterisk is open source, you can make the
unbelievably simple change yourself.  If you make it configurable and
default it to "no" (so as not to break backwards compatibility, not to
mention RFC compliance), it may even get accepted into Asterisk so
that you won't have to maintain your own patchset.

This feature would actually be a bit like "alwaysauthreject" in that
it breaks RFC compliance for the sake of security, so it's not a
completely lost cause.  However, pining away on a mailing list about
how simple the work would be instead of doing it yourself is.

Regards,

Matthew Roth

=
And, I suspect someone somewhere would do it for a $25 bounty, and the
original poster could save $50,000 in telecom billings and email time.

:-)
CF


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Re: [asterisk-users] Anybody doing PRI over IP?

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 07, 2011 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anybody doing PRI over IP?

- Original Message -
> There is a "T1 over Ethernet" scheme that runs a T1 over Ethernet, Up
> all the time. It consumes 1.5 +/- megs 24/7.
> 
> I would suspect that was what was being offered.
> 

Wow, that sounds horrifically inefficient. *MAYBE* keep the D channel open
all the time if necessary, but what about all the channels that are not in
use? Just waste 64k for each channel just because?

Do you have any information on this technology, or the name of the vendor
that offers it? You've piqued my interest. :)

--Tim

The reason for this scheme as proposed to us was for FAX use rather than
other means.

The people who proposed it however within a week converted to "Plan B".  So,
I have no further info.

CF


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Re: [asterisk-users] Anybody doing PRI over IP?

[Cary Fitch]

There is a "T1 over Ethernet" scheme that runs a T1 over Ethernet, Up all
the time.  It consumes 1.5 +/- megs 24/7.

 

I would suspect that was what was being offered.

 

Cary

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, July 07, 2011 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anybody doing PRI over IP?

 

  _  

From: "eric weaver" 
Sent: Thursday, July 07, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Anybody doing PRI over IP?

A carrier I like will be introducing PRI over IP, presumably going thru some
sort of gateway box (I'm guessing by Adtran but no data yet).  Has anybody
set up successfully to work directly with such a feed without bothering to
take it down to T1 and use  a T1/PRI card?

Thanks

eric 

I agree with others that likely what you are getting is a product that is
SIP based and it is just being priced and bundled to compete with a PRI
connection as most bussiness owners and phone guys know what a PRI is..  We
have pri's into gateways that run on our VOIP network and we have sip trunks
and we mix services out to our customers based on what the routes require.
Most of our up line CLEC's can now deliver their TDM and SIP services in
both forms so in most cases we take the SIP version and where the vendor
does not support SIP correctly we take their PRI version and convert it to
SIP ourselves on our gateways.  

zktech



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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[Cary Fitch]

-- 
I know we've come a long way,
We're changing day to day,
But tell me, where do the children play?

Yusuf Islam, Nov. 1970

AKA CAT STEVENS.

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Re: [asterisk-users] PRI issue its BUSY

[Cary Fitch]

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, June 06, 2011 8:20 PM
To: asterisk-users
Subject: [asterisk-users] PRI issue its BUSY

 

Hi all,

I just configures my PRI and incoming calls are working fine but outside
calling giving error PRI is BUSY :(  any idea ?  I have same setup on other
box and that boxes works perfect.

-- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002
-- DAHDI/i1/6463279153-2 is making progress passing it to
SIP/7328-0002
-- DAHDI/i1/6463279153-2 is busy
-- Hungup 'DAHDI/i1/6463279153-2'
  == Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY'

 

Maybe the problem is external to the box.

 

Try swapping PRIs briefly for testing.

 

C.

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Re: [asterisk-users] asterisk fails when DNS or internet fails

[Cary Fitch]

It seems to me:

That every Asterisk system that is being used for PBX or other "internal" as
well as external use should have a local DNS, run either on the same box or
on an adjacent box.

A simple BIND installation is low overhead.  If remote phones use it for DNS
then if they are on the net, they have all info they need to make a call
anywhere, and commonly called locations are either in DNS or cached.

Then there are no "DNS" based failures unless the Net is down or the
Asterisk box is down. 

Well, now, I should do as I preach and install Bind on our Asterisk box.
However since we do have multiple DNS locally, we are effectively following
this advice.

C.


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Re: [asterisk-users] Free CNAM

[Cary Fitch]

> The system uses real Telco CNAM Dips.  Any generic names you get are 

> from the subscriber's carrier itself.  We can only provide what we 

> ourselves get.

 

There's more than one CNAM database (aren't there seven?).  I would have

hoped that a service such as this would look at a bunch of them and choose

the one that had the best result.

==

I am the original skeptic in this thread. (and, horrors, have top posted
besides!)

 

But, 

 

What do you want for free?

 

Oh, Asterisk for starters of course, but... don't look a gift horse in the
mouth.  Say "Thanks, I hope it grows."

 

So how about some one in the open source community come up with a few lines
of code to add it to Asterisk?

 

C.

 

 

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Re: [asterisk-users] Free CNAM

[Cary Fitch]

GRIN!
C.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Sunday, May 29, 2011 8:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Free CNAM

On 05/29/2011 09:37 AM, Michael R. Wally wrote:

So how long till its an adaptive telemarketing blocker based on the 
query velocity of the numbers ?



> The system uses real Telco CNAM Dips.  Any generic names you get are 
> from the subscriber's carrier itself.  We can only provide what we 
> ourselves get.
>> I tried it, but it returns the same kind of junk that some of the 
>> databases
>> do.  For example, on a Florida number, it just says "FLORIDA" instead of
>> the proper name (some of the CNAM databases have the right name).
>>
>
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Re: [asterisk-users] Free CNAM

[Cary Fitch]

Thanks I had just sent off a mea culpa, and I have a gold plated tin foil
hat that I love and will continue to wear!

:-)

Cary

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Sunday, May 29, 2011 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Free CNAM

You are doing a CNAM lookup on that 202 number.  Change the URL to a
number you know, and it will do a CNAM lookup on it.  You can take your
tinfoil hat off now.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Sunday, May 29, 2011 8:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Free CNAM

BE WARY OF THIS ONE!

If you click the link it comes up with a simple block Text Message  
"US GOVERNMENT"

I doubt the US Government has any thing to do with it but... something
is
fishy here. 

Cary



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R.
Wally
Sent: Sunday, May 29, 2011 6:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Free CNAM

FreeCNAM.org is providing a free CNAM API for Open Source PBX users.
This API queries a private CNAM database, and returns standard
15-Character CNAM results. Any entry not already in the database will
be queued for investigation, and added to the database as soon as
information is located. This system has access to several CNAM
backends, and is not a party to any use-limiting or no-caching
agreements.

The API is: http://freecnam.org/dip?q=2024561414

You can monitor the stats, including the current queue size, at
freecnam.org

API Results will continually improve as the database grows, so please
be patient with limited results at this early stage.

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Re: [asterisk-users] Free CNAM

[Cary Fitch]

Excuse me!

Doing that with  http://freecnam.org/dip?q=9038874180 

Comes up with WAL-MART which is correct, so I guess my mistrust was
paranoid.  (But just because you are paranoid doesn't mean that someone
isn't trying to infiltrate your computer or steal your email address or what
ever!)

And the returned page is perfectly clean as far as HTML code is concerned!

I DO suggest changing the example so as not to alarm suspicious minds.

Cary

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R.
Wally
Sent: Sunday, May 29, 2011 6:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Free CNAM

FreeCNAM.org is providing a free CNAM API for Open Source PBX users.
This API queries a private CNAM database, and returns standard
15-Character CNAM results. Any entry not already in the database will
be queued for investigation, and added to the database as soon as
information is located. This system has access to several CNAM
backends, and is not a party to any use-limiting or no-caching
agreements.

The API is: http://freecnam.org/dip?q=2024561414

You can monitor the stats, including the current queue size, at freecnam.org

API Results will continually improve as the database grows, so please
be patient with limited results at this early stage.

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Re: [asterisk-users] Free CNAM

[Cary Fitch]

BE WARY OF THIS ONE!

If you click the link it comes up with a simple block Text Message  
"US GOVERNMENT"

I doubt the US Government has any thing to do with it but... something is
fishy here. 

Cary



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R.
Wally
Sent: Sunday, May 29, 2011 6:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Free CNAM

FreeCNAM.org is providing a free CNAM API for Open Source PBX users.
This API queries a private CNAM database, and returns standard
15-Character CNAM results. Any entry not already in the database will
be queued for investigation, and added to the database as soon as
information is located. This system has access to several CNAM
backends, and is not a party to any use-limiting or no-caching
agreements.

The API is: http://freecnam.org/dip?q=2024561414

You can monitor the stats, including the current queue size, at freecnam.org

API Results will continually improve as the database grows, so please
be patient with limited results at this early stage.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Cary Fitch]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, May 04, 2011 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

 

 

2011/5/5 Flavio Goncalves 


but stuffing Asterisk with
many  new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.


yes but it seems to me that code stability is improving.
Maybe next 1.10.0 version will be "production-ready" from day 1 ?
 


Flavio E. Goncalves
www.asteriskguide.com



 

Compare to which version of Windows. Patches are a never ending process

 

Cary Fitch

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Re: [asterisk-users] Occasional call from "asterisk"

[Cary Fitch]

We were getting "a lot" of those. We installed IPTables with blocking of
everything outside of North America and they all but vanished.

No direct evidence, but a pretty good empirical guess that they were related
to hackers trying to get paths to the US.

CF

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning
Sent: Thursday, April 07, 2011 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Occasional call from "asterisk"

Hi,

Now and then our SIP phones ring with "asterisk" showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup).  Can anyone offer some insight?  Here's
relevant snippets from my extensions.conf and Master.csv log:

This line shows up in Master.csv:

"","","1-NOANSWER","inbound","","DAHDI/1-1","SIP/505-0150","Dial","SIP/5
01&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr","2011-04-07
21:37:05","2011-04-07 21:37:16","2011-04-07
21:37:21",16,5,"ANSWERED","DOCUMENTATION","1302212225.444",""

Here's [inbound] from extensions.conf:
[inbound]
exten => s,1,Answer
exten => s,n,Ringing
exten => s,n,Set(CALLERID(num),9${CALLERID(num)})
exten => s,n,Dial(SIP/504&SIP/506,5,tTgr)
exten => s,n,Goto(1-${DIALSTATUS},1)
exten => 1-ANSWER,1,Hangup
exten =>
_1-.,1,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr)
exten => _1-.,n,Goto(2-${DIALSTATUS},1)
exten => 2-ANSWER,1,Hangup
exten => _2-.,1,Voicemail(499@default,u)
exten => _2-.,2,Hangup

The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings.  Works great most of the time.

I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there.  (The
preceding line of the log is a call from almost an hour earlier).  In that
case though I'd expect to see "1-CANCEL" in the log instead.  Perhaps if the
caller happens to hang up right between the two Dial() commands?..

As an aside, the Set(CALLERID...) bit doesn't work.  The idea was to prepend
a 9 so that a SIP user could use the "redial" feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line).  Unfortunately the 9 does not get prepended.

Thanks in advance for any and all advice!
~Brian

-- 
  Brian Henning, Software Engineer

/\Pine Research Instrumentation 
   //\\   5908 Triangle Drive 
  ///\\\  Raleigh, NC 27617 
  USA 
|| 
||phone: 919.782.8320 
  fax:   919.782.8323 
  email: bhenn...@pineinst.com 
-- 



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Re: [asterisk-users] asterisk and fail2ban

[Cary Fitch]

You are a bad person! ;-)

CF

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, March 31, 2011 10:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk and fail2ban

On Thu, Mar 31, 2011 at 10:42:52AM -0500, JR Richardson wrote:

> I have F2B set to ban after 1 attempt.  The most I have seen in the
> logs is 4-5 attemps before ban is applied.  I am calling scripts that
> apply the ban to a cisco access-list, so there is script/telnet/config
> delay but it is very minimal and works very well.

So I forge one SIP packet and I get you to block the IP address of your
SIP trunk (or your IAX trunk)?

Cool!

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] asterisk and fail2ban

[Cary Fitch]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, March 31, 2011 10:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and fail2ban

 

On Thu, Mar 31, 2011 at 7:17 AM, vip killa  wrote:

Back to the original question, for those of you using Fail2Ban,

Does it take an unusually high amount of break-in attempts before attackers
are banned?

I have it set to 5 attempts in fail2ban but usually, the attacker is able to
make over 100 attempts before fail2ban bans them.

I've tried this using asterisk's /var/log/asterisk/messages and
/var/log/messages with same results.

Perhaps someone else is experiencing this or has resolved it, thank you.

 


Check your log files.  With the current generation of SIP attack scripts,
I've seen hundreds of attacks come in within one second, especially if
you've got decent bandwidth.  I've seen fail2ban logs that state between
60-250 failed attempts for asterisk.  I think it's just the nature of the
speed of the attacks.  

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com

 

 

Which is a good reason to use manual, mass IPTables entries for "the rest of
the world" and fail2ban generated entries for creeps in your
neighborhood/country.

 

CF

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Re: [asterisk-users] asterisk and fail2ban

[Cary Fitch]

Obviously, the other side of the world wants connections to your side, no
matter what side you are on.
:-)

Cary


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, March 29, 2011 3:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk and fail2ban

Le 29/03/2011 19:34, Sherwood McGowan a écrit :
> On 3/29/2011 12:25 PM, Steve Edwards wrote:
>>> On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
 First thing I'd do is restrict the ip blocks your sip endpoints can
 register/call from in sip.conf (or your database's table for sip
 endpoints)
>> On Tue, 29 Mar 2011, Gilles wrote:
>>
>>> Thanks for the idea, but it's not possible, as the Asterisk must be
>>> accessible for road warriors and receive SIP calls from anyone.
>> Really? How many callers are you expecting from North Korea, Libya,
>> China, Iran, etc?
>>
> Thanks Steve, you just emailed exactly what I was going to say...
>
> Remember guys, there's a LOT of IP blocks out there that are almost
> definitely not going to be somewhere you expect to receive SIP traffic
> from.

Well, I can tell you that our servers in europe those days are mainly 
attacked by US IP ranges (remember last year the problem with amazon 
cloud). They now disappear here in europe but lots of other US networks 
quickly replace them :-(

-- 
Daniel

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Re: [asterisk-users] Caller ID

[Cary Fitch]

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, March 01, 2011 11:31 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Caller ID

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Tuesday, March 01, 2011 11:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Caller ID

 

We do not get caller ID (name) on our telco lines.

 

However we have a few single line extensions with consumer type handsets
that ring at odd hours with "Asterisk" before the phone is picked up, and
"Out of Area"  after it is picked up.

 

I have read that "Asterisk" is what is reported by Asterisk for 0 length
caller ID number.

 

But since we don't subscribe to Caller ID Name, I am wondering where the
"Out of Area" is coming from?

 

Could these be hacking attempts via IP?   Perhaps they are doing the caller
ID name?  It only happens to a few extensions as far a we know.

 

TIA for any input or knowledge.

 

IP Hacking "should not" apply on your Telco lines.  I'd start with your CDR
file (/var/log/asterisk/cdr-csv/Master.csv) and go from there.

 

 

I had in mind a hacking attempt IP calling extension lines that exist, but
thanks, I will look at the cdrs.

 

 

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[asterisk-users] Caller ID

[Cary Fitch]

We do not get caller ID (name) on our telco lines.

 

However we have a few single line extensions with consumer type handsets
that ring at odd hours with "Asterisk" before the phone is picked up, and
"Out of Area"  after it is picked up.

 

I have read that "Asterisk" is what is reported by Asterisk for 0 length
caller ID number.

 

But since we don't subscribe to Caller ID Name, I am wondering where the
"Out of Area" is coming from?

 

Could these be hacking attempts via IP?   Perhaps they are doing the caller
ID name?  It only happens to a few extensions as far a we know.

 

TIA for any input or knowledge.

 

 

 

 

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Re: [asterisk-users] [zapata.conf] What is "wink"?

[Cary Fitch]

Wink, I think is a "start" protocol aks "wink start".  It is like a flash,
but happens as part of the predialing/dialing process.  

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Tuesday, March 01, 2011 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [zapata.conf] What is "wink"?

 

http://www.carrieraccessbilling.com/telecommunications-glossary-w.asp


 

  _  

From: asterisk-users-boun...@lists.digium.com on behalf of Gilles
Sent: Tue 3/1/2011 7:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [zapata.conf] What is "wink"?

Hello

I couldn't find information about what "wink" is in zapata.conf:

www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters

Does someone know what it is, and how it differs from flash?

Thank you.



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Re: [asterisk-users] Google Voice outbound Caller ID broken

[Cary Fitch]

What kind of "broken" are you seeing.

 

It could be the ID is pseudo ID and may never reflect the actual caller.

 

CF

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Thursday, February 24, 2011 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Google Voice outbound Caller ID broken

 

Anybody else noticed that caller id for outbound calls via Google Voice
seems to be broken?  It seems to be a Google Voice problem though, not an
asterisk issue.

-- 
Chris

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[asterisk-users] "Asterisk" caller ID

[Cary Fitch]

We are getting a lot of calls identified as "Asterisk" or "out of area" in
the middle of the night.

 

>From other posts on the list, I have assumed these are null Caller ID calls
and Asterisk is plugging in pseudo ID.   Is that correct?

 

It seems to me that Asterisk should simply say "no caller ID" or "No ID" or
something besides "Asterisk".

 

In any case, we are trying to filter them with little success.

 

When we do a LEN(CALLERID(num) we get "13", when we expect "10"

 

The call pattern is 1 call followed by a second abut 1 minute later followed
by 1 about 10 minutes later.

 

Does anyone have any ideas to contribute?

 

Thanks

 

Cary

 

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Re: [asterisk-users] REFER and dialplan broken(asdocumentedinchan_sip.c on line 11951)

[Cary Fitch]

It is free if you can use it.  You can pay for all the help you want to or
have the money to pay for.

 

The "Asterisk Software Charity Society" went bankrupt about 2500 years ago. 

 

You can pick some name from the mail list and demand they fix the issue you
perceive.  But you probably won't be able to wait for results.

 

There are 3 legs to any transaction.  Speed, Quality, Price.   You get to
pick any two.  The other party gets to set the 3rd one.

 

You can't set all 3.

 

Cary

 

Except in Wisconsin.

 

Even there.

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Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951)

[Cary Fitch]

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as
documentedinchan_sip.c on line 11951)

 

I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.

It is free if you can use it.  You can pay for all the help you want to or
have the money to pay for.

 

The "Asterisk Software Charity Society" went bankrupt about 2500 years ago. 

 

You can pick some name from the mail list and demand they fix the issue you
perceive.  But you probably won't be able to wait for results.

 

There are 3 legs to any transaction.  Speed, Quality, Price.   You get to
pick any two.  The other party gets to set the 3rd one.

 

You can't set all 3.

 

Cary

 

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Re: [asterisk-users] On-Hold Music

[Cary Fitch]

And there is also ASCAP:  American Society of Composers, Authors and
Publishers.

Also a smaller than either group: SESAC.

Cary 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, February 14, 2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] On-Hold Music

On Monday 14 February 2011 08:23:08 Danny Nicholas wrote:
>  Might not be your question to answer, but if I did get a BMI 
> license, this would allow me to use "virtually any music I wanted" for 
> MOH?

The answer is, as long as the music publisher for each piece of music has an
agreement with BMI to license their music, yes.  You can verify each
individual title here:
http://www.bmi.com/search/

--
Tilghman

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Re: [asterisk-users] Top Posting

[Cary Fitch]

 
> How amusing that you follow that statement by being too lazy to trim
> all of the irrelevant crud after your comment by pressing
> ctrl-shift-end followed by delete. It works in Outlook.  
> 
> Tom

This is the problem, everyone has a personal goal.  One side wants fast
replies at the top, with no interest in the repetitive, redundant
signature/disclaimers & content below.  The other wants total historical
readability or questions and answers in top to bottom readability in every
message.  And, this is a type of list that is used by 1000s of individuals,
not people from a single company.  We are just lucky we don't have someone
posting in sentences that read from right to left. :-)

Also most (all?) mail clients don't allow setting preferences based on the
source of the message.  I.E. Top post for email, bottom post for the cooking
list and bottom post for the Asterisk list. 

And then almost no one trims anything no matter what their
preferences/beliefs are, and yells at others for top or bottom posting or
interleaving, usually while violating some other list rule or general net
etiquette.

How about just no quoting or only the actual last message you are replying
to?  The list doesn't require any quoting. Contribute your thoughts, and
leave it at that. Everyone has the previous posts on their computer, if they
don't know the history, let them go back and read.

Cary


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Re: [asterisk-users] Top Posting

[Cary Fitch]

Paul Belanger wrote:
> Moderation would be another option (personally opinion). Regardless,
> we should all now be aware of the rules [1] of the mailing lists. 
> All we can do now is hope people respect them.  
> 
> [1] http://www.asterisk.org/community/rules
> 
> --
> Paul Belanger
> Digium, Inc. | Software Developer


With that type of trimming and my own trimming, bottom posting works for me,
as well as top posting.  There is little difference.

But with 5 screens of text, , 7-10 repeated messages multiple signature
lines and other tripe, bottom posting is a PITA.

So if others trim, I am happy to bottom post.

Cary Fitch


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Re: [asterisk-users] Top Posting

[Cary Fitch]

> 
> Hi,
> 
> I thought this kind of discussion didn't exists in asterisk list.
> I guess most tech list members will argue on top-vs.-bottom subject at
> some point :) 
> 
> anton

It only exists when someone starts the discussion.  The top posters never
start it.

Bottom posting wouldn't be bad if the posters trimmed all but the message
they are replying to from the reply.
(See what I did?)  Top posting puts the recent replay clearly as the first
thing on the screen.  

I often have messages with 5 to 10 screens of previous posts to wade through
to get to the bottom.

If "repliers" would trim old content from their replies, it would save a lot
of readers time and make bottom posting very efficient.

I went through this type of discussion 20-25 years ago because we were all
on 2400 to 14,400 baud modems, and trimming was demanded.

Then Rich Text and HTML came along and you should have heard the screaming
about bloated messages.  Six-ten line signatures with ascii-art were not
really appreciated either.

Cary Fitch







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Re: [asterisk-users] Top Posting

[Cary Fitch]

Tom Rymes wrote:
> On Jan 15, 2011, at 9:29 AM, Don Kelly wrote:
> 
>> That said, of course I want to follow this list's etiquette. I've
>> posted a couple times asking how I can interleave responses in
>> Outlook or what other approach can I take to make it practical to
>> stop top-posting. Any suggestions?
> 
> Don:
> 
> Outlook-QuoteFix:
> http://home.in.tum.de/~jain/software/outlook-quotefix/ 
> 
> I found that program last night after reading one of the pages linked
> in this thread. The program isn't supported on OL 2007 and newer, but
> there is a link on the page to a macro for newer versions. Wish I had
> known about it years ago!   
> 
> Also, http://mailformat.dan.info/config/outlook.html shows the
> general steps needed to make Outlook approximate standards. 
> 
> HTH,
> 
> Tom


Thanks.
As far as making bottom posting work, >
http://home.in.tum.de/~jain/software/outlook-quotefix/ makes it so much
simpler and better!

I just installed it, and will try to keep using it.  If I don't want the
previous material at the top, I will delete it. But, personally, I really
prefer top posting or the previous material deleted.

However... Time will tell.

Cary




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Re: [asterisk-users] sip attack.. fail2ban not stopping attack

[Cary Fitch]

Simply to reduce the attack, and then improve the defense:

If you don't need traffic from some area that is attacking you, just put the
whole area in IPTables.  A list is available on VOIP-INFO.org.

Cull out what you want to allow.

Then tune Fail2Ban at your leisure.

Cary Fitch



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Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, December 22, 2010 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer
45KSouth London

On Wed, 2010-12-22 at 11:23 -0500, C. Savinovich wrote:
> 
> 45K ?
> 
> With 45K I can barely pay for gas, tolls, and breakfast.  If you guys
> are such a fast growing company, probably you can pay better salaries.
> 
> CS
> 

And you have to know "Kalamino"!  :)

You know "Kalamino"?  I haven't seen him since prison in Budapest! Tell him
hello!   :-)

CF


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Re: [asterisk-users] Maximum E1 Ports on Asterisk ?

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Wednesday, December 22, 2010 6:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ?

On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo
 wrote:
> Hi All,
>
>
>
> Just a little over thought. Sorry if someone already asked about this
> before.
>
>
>
> Is it possible to put all 16 Ports of E1 in One Asterisk Server ?
>
>
>
> And if it's not possible is there any suggestion or alternative for me to
> use more than 320 lines of outgoing calls on One Asterisk Server ?
>
>
>
> Thanks
>
>
>
> ZH


Zoel

It is possible to do what you are asking. In general the issue is
raised about having "all your eggs in one basket" where one server or
hardware failure can drop all of your lines for a period of time.
External solutions like Xorcom and Redfone are great ways of
abstraction.  The concurrent call load on a server relies on the work
to be done on each call.  If you are using multiple codecs and
recording the calls in another file format with other complex dialplan
or AGI scripts then one server may not handle the calls well.  If
everything is ALAW and just dialing though then this would not be a
problem for one server.  If you search the list for "sizing"
"concurrent" and "load" you will find more information.  One very nice
thing is that testing is very easy with or without the E1 hardware,
try running the TDMoE channels between two servers and run a SIPp or
other test to see the issues in a lab.

~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

--
_

In a previous post I also mentioned Xorcom.  They do have a unique fail over
ability with their Astribank systems. 

With dual servers, separate chassis and power supplies for the 4 port T1/E1
cards, USB interconnections, and redundant power supplies for the
Astribanks, system downtime can be minimized, and if there is a failure,
repair would be at worst, no screwdriver needed.  

If system failure would be idling 200 - 400 people, avoiding system down
time would be a major objective.

Cary Fitch


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Re: [asterisk-users] Maximum E1 Ports on Asterisk ?

[Cary Fitch]

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zoel Hairi -
Yahoo
Sent: Wednesday, December 22, 2010 5:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Maximum E1 Ports on Asterisk ?

 

Hi All, 

 

Just a little over thought. Sorry if someone already asked about this
before.

 

Is it possible to put all 16 Ports of E1 in One Asterisk Server ? 

 

And if it's not possible is there any suggestion or alternative for me to
use more than 320 lines of outgoing calls on One Asterisk Server ?

 

Thanks 

 

ZH

 



 

The general answer is "Yes, maybe".  I suggest you look at the Xorcom.com
website for their load test data.  Using a well sized server with "best
practice" tweaks is important.

 

It appears that bigger is not always better.  For instance it seems to hurt
or at least give no benefit to use a quad core processor.  We just ran tests
that indicates  

Xorcom's  "3000" model would handle 16 CAS T1s.

 

CAS T1s produce a very high interrupt rate.  PRI T1s don't cause nearly as
high rate.  The right choice of cache, motherboard, processor and "tweaks",
are essential.  You would be "leading the pack". I think.

 

.Cary Fitch

 

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[asterisk-users] Asterisk as a caller ID

[Cary Fitch]

In a 1.4.24 system, out of several lines, one of ours gets 1 or more random
calls a day with "Asterisk" as the caller ID.

I have just seen this described in the last couple of weeks, but at the time
it wasn't happening to us, and I the explanation didn't stick with me.

Can anyone give me a pointer to this "feature"?  Searching the message base
for "Asterisk" seems futile.

Thanks!

Cary Fitch


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[asterisk-users] Spam

[Cary Fitch]

I have been pounded with new, mostly text spam in the last few weeks.
Tonight I realized that the address that is being spammed is a personal one
I use for this list.

Has anyone else noticed "new spam" in the last 2-3 weeks?

Cary Fitch


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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[Cary Fitch]

I am not sure where you are and what legal conventions are involved.

 

Are you saying the Telco (and legal restrictions) say you can’t send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the “logic” in that?

 

Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? 

 

Or are you concerned about Asterisk handling the TDM to IP conversion in an
adequate manner?

 

I am not sure/aware myself that Asterisk will do a modem to IP conversion.
I think in your example the AS5300 is doing that.

 

What is the Telco’s problem in doing what the customer was doing before?

 

Feel free to correspond directly if you want to.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo
Méndez Soto
Sent: Wednesday, November 24, 2010 7:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??

 

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet

Ideal topology:
Telco --> SS7 --> TE410P-AsteriskServer --> Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
<mailto:asterisk-users@lists.digium.com> 

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


José Pablo Méndez
  

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Re: [asterisk-users] GSM and SS7 Questions

[Cary Fitch]

In regard to #2, "any" T1 card should work.  But the problem is you need SS7
software and SS7 connectivity in addition to the T1 card.

Cary Fitch








-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Sent: Wednesday, November 17, 2010 2:31 PM
To: asterisk-users
Subject: [asterisk-users] GSM and SS7 Questions

I have two questions for the group.

#1 - I'm looking to use some GSM SIM cards with my Asterisk PBX.   Can
anyone recommend a gateway?  I need about 10-15 SIM slots.

#2 - I'm also looking to connect Asterisk to an SS7 signaled DS1 (24
channels) for inbound and outbound voice calls.  Can anyone offer any
suggestions for cards to use there?

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Re: [asterisk-users] T1 with Robbed Bit Signaling

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, November 16, 2010 2:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 with Robbed Bit Signaling

How many T1s per server?

And, the T1s would be DS0s from customer's homes.

Cary






> We are trying to get DS0 EELs from the Telco via T1s and are not sure what
> their trunk scheme will be. We are guessing it will be RBS T1s with DTMF
> from the customer premises.

To connect two remote offices?  Why don't you use SIP/IAX over the net or 
over a tunnel?


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Re: [asterisk-users] T1 with Robbed Bit Signaling

[Cary Fitch]

So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber.

How does the Telco send you the called party info and you to them? DTMF?
What signaling system is being used? E&M or?

We are trying to get DS0 EELs from the Telco via T1s and are not sure what
their trunk scheme will be. We are guessing it will be RBS T1s with DTMF
from the customer premises.

To enhance the question:  Is anyone on list getting customer loop EELs from
a US (AT&T) Telco, and if so, what flavor and other details?

Cary Fitch


boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere

On Tue, 16 Nov 2010, Cary Fitch wrote:

> Has anyone here used T1s with RBS with asterisk?
>
> Cary Fitch
>

All of my T1s are RBS.  No PRI service here.  Am using both Digium and 
Sangoma cards.
LaCoursiere

So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber.

How does the Telco send you the called party info and you to them? DTMF?
What signaling system is being used? E&M or ???

We are trying to get DS0 EELs from the Telco via T1s and are not sure what
their trunk scheme will be. We are guessing it will be RBS T1s with DTMF
from the customer premises.

To enhance the question:  Is anyone on list getting customer loop EELs from
a US (AT&T) Telco, and if so, what flavor signaling,  and other details?

Cary Fitch

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[asterisk-users] T1 with Robbed Bit Signaling

[Cary Fitch]

Has anyone here used T1s with RBS with asterisk?

Cary Fitch


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Re: [asterisk-users] How to construct a call center on asterisk

[Cary Fitch]

Sorry, I don't have any experience in the call center area.

 

Cary

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Monday, November 15, 2010 10:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to construct a call center on asterisk

 

Thanks Cary,

I constructed IVR (Interactive Voice Response) system on asterisk and now i
want to construct a call center to support IVR system.The call center that i
want to construct has 7-8 employees so that i don`t want to buy it. I found
on internet and guided using Open SIP Server with asterisk to construct a
call center but i don`t know how to do now. Do you know using Open SIP
Server with asterisk to do?please help me!
Thanks and best regards.

Phuong Hoang.

On Tue, Nov 16, 2010 at 10:32 AM, Cary Fitch  wrote:

That is a broad question.

 

If it is a 10-20 person call center, you may do it.

 

If it is a high density center with lots of lines, requiring fail over
capability, etc. so that hundreds of employees are not sitting around during
down time,  I would suggest "designing it" by buying it from a manufacturer
that has done it before.  I know of a company whose name starts with "X" .
There may be others.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Monday, November 15, 2010 8:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to construct a call center on asterisk

 

Hi all
Now, i want to construct a call center on asterisk but i don`t know how to
do. Can anyone help me?
Thanks and best regards.


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Re: [asterisk-users] How to construct a call center on asterisk

[Cary Fitch]

That is a broad question.

 

If it is a 10-20 person call center, you may do it.

 

If it is a high density center with lots of lines, requiring fail over
capability, etc. so that hundreds of employees are not sitting around during
down time,  I would suggest "designing it" by buying it from a manufacturer
that has done it before.  I know of a company whose name starts with "X" .
There may be others.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Monday, November 15, 2010 8:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to construct a call center on asterisk

 

Hi all
Now, i want to construct a call center on asterisk but i don`t know how to
do. Can anyone help me?
Thanks and best regards.

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Re: [asterisk-users] Big practical systems

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak
Sent: Sunday, November 07, 2010 2:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big practical systems

I believe this looks like a standard channel bank.  Asterisk generates all
audio.  Ring and hook status are sent out of band.  Dial tones are in-band.
Ringback, busy, congestion are in-band audio.  I would think a standard T1
card would be fine.

That said, I would verify this with the LEC. 
===

Does anyone know if AT&T EELs delivered to a CLEC would be PRI, or Robbed
bit?

Cary Fitch 


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Re: [asterisk-users] Big practical systems

[Cary Fitch]

Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into PRIs.

> Thanks, will look at that.  Ah, a DS3/T1 mux.  I was looking for a DS3
PC Card... it would have 672 "channels" but the system doesn't need to
handle but "<20%" of them at one time.

If you're just talking 150 calls, you could do that with two 4x port
cards in a single PC. I thought you were talking a lot bigger.

==>I mean DS3 with 672 channel paths. There are 672 subscribers "out
there".  I am saying that only a percentage of them are talking at peak
times.  We need to "supervise" 672 lines and expect 15% to talk at the same
time.

> Each line/call would have to have absolute caller ID.  In other words,
PSTN
> call handling.

Ummm, there's no such thing as absolute caller ID. You wanna try that
question again? callerID is not legally binding, is not used by
billing, anybody can spoof it.

===>I mean we have to provide service and know what line is calling, not
just provide anonymous service to a lot of people.  We can't just mux a
bunch of lines in to the Asterisk box with no identification.


Cary


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Re: [asterisk-users] Big practical systems

[Cary Fitch]

Alternate question:

Asterisk/PSTN oriented.

If an Asterisk system were interfaced via a T1 to a local telco loop to a
customer premises:

(This is not a T1 to the customer premises, but a T1 to the telco who then
demuxes it to copper to the customer premises.  IE. In Telecom terms an
EEL.)

Will Asterisk handle that scenario with common drivers and cards?

Who generates the customer audio "comfort sounds", ringing, busy, etc?



Cary
"I know a lot, but not everything."


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Re: [asterisk-users] Big practical systems

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benoit
Sent: Sunday, November 07, 2010 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big practical systems

Hi,

Did you saw this before:
http://lists.digium.com/pipermail/asterisk-users/2008-April/209146.html
?

-- 
---

THANKS, YOUR ANSWER IS CONTRUCTIVE.

No, I had not seen that.  But it is helpful and gives me the idea that I
should look at 4 systems and/or about 7-8 T1s per system maybe.

That has "failure-spreading" and localization benefits also. As well as not
trying to handle a whole LATA or more in a box.

We still have a lot of practical matters to work out.

Cary


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Re: [asterisk-users] Big practical systems

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Sunday, November 07, 2010 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big practical systems

inline

> I don't want to start the "How many calls can Asterisk handle?" discussion
> or "How many angels can stand on the point of a pin?" discussion either.

You just did

Nah.

> But can anyone contribute some practical knowledge of systems that take in
> channel bank T1s or DS3s from "far away", and process the calls?

Most of us are far too busy.  These things are best learned the hardway.

Ha, ha, ha, ha, ha.  My Lucent 5010 Compact Switch with DS3 connections to 8 
LATAs keeps me moderately busy but I still have time for Asterisk.  I have 
learned everything the hard way.

> I am looking for real world, been there, done that, or "check the 'Belchfire
> Systems GigaFiber 65536' system".

Its called Asterisk.
===
Thank you, and so you've been there and done that but don't want to help, just 
be cute?

> Not to start the discussion, but "Is there a board that will take a DS3 (672
> channels) and a system that will handle the calls, or is that a silly
> question?"

There are many.  The primary problem it getting a provider to provide
you with a DS3.

=
Obviously I don't have that problem, and didn't ask for help with that.

> Is there an IP box that would take the DS3 and then a system that would
> handle the calls? My guess would be yes because the actual call load would
> be far lower than 672 calls.  Maybe 100-150 or so simultaneous.

There are a few solutions here and several expensive chunks of
hardware. Do you want to put all your eggs in one basket?


===

No, There are many eggs and can be many baskets.

> Each line/call would have to have absolute caller ID.  In other words, PSTN
> call handling.

That is between you and the provider.  The technology exists on the wire.

===
Good, that is why I as asking for pointers.


> Cary

Gringo Malvado...

==
I know what "Gringo" means, and I suspected "malvado" but looked it up.  From 
Wiktionary.org:

Adjective
malvado m (feminine malvada plural malvados feminine plural malvadas; 
comparable)
bad; evil; wicked
[edit]Inflection
[show ▼]  Inflection of the Portuguese adjective malvado
[edit]Spanish

[edit]Adjective
malvado m (feminine malvada, masculine plural malvados, feminine plural 
malvadas)
evil, wicked, mean, bad
[edit]Synonyms
perverso
[edit]Related terms
maldad
maleado
malo

And you are a regular poster here? Muy malo.

Cary 



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[asterisk-users] Big practical systems

[Cary Fitch]

I don't want to start the "How many calls can Asterisk handle?" discussion
or "How many angels can stand on the point of a pin?" discussion either.

But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from "far away", and process the calls?

I am looking for real world, been there, done that, or "check the 'Belchfire
Systems GigaFiber 65536' system". 

Not to start the discussion, but "Is there a board that will take a DS3 (672
channels) and a system that will handle the calls, or is that a silly
question?"

Is there an IP box that would take the DS3 and then a system that would
handle the calls? My guess would be yes because the actual call load would
be far lower than 672 calls.  Maybe 100-150 or so simultaneous.

Each line/call would have to have absolute caller ID.  In other words, PSTN
call handling.

Cary




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Re: [asterisk-users] Why are the hackers scanning for these?

[Cary Fitch]

Adding on more thoughts:

Think what Google has done in Mapping the Earth, Mapping the Web, and now
working on Google Voice and Google Mail.

Every one of those makes money either directly and/or synergistically with
other components.

Now consider someone with "telephone" interests or spam interests.  In this
modern database and filtering and probing age, load in ARIN or RIPE IP
Ranges, start building database data and filters, and let it run...

And the other IP areas too.

Cary


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Re: [asterisk-users] Why are the hackers scanning for these?

[Cary Fitch]

I've just switched my outbound ip address a week ago. Not static, but
dhcp on TimeWarner cable.  I've registered only with another of our
offices. The outbound calls are all pstn bound through Teliax.

But somehow my log is filling up with registration requests over this
new ip address from a bunch of addresses. How can these guys find my
new ip address? Or are they just scanning all ip addresses in
creation?

sean

-- 
_

"Follow the money"

Just like for Spam, there is money in Sip-Hacking.

Anyone that has SIP traffic to move (selling the service) has money.  If
they can move it for free, even more money.  A few servers running Hacking
programs (SIPVicious) or e-mail server hacking programs is no big deal and
bandwidth at colo centers is unlimited.

Then they convert to BOT controllers and have free computers and bandwidth
world wide.

They generate a database of public IP addresses (DHCP, whatever) and have a
target of poorly protected IPs to troll.

"Lucky you". ;-)

Cary




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Re: [asterisk-users] Why are the hackers scanning for these?

[Cary Fitch]

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, November 07, 2010 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why are the hackers scanning for these?

 

> Here's some examples:

>2648061411
>3190339404

I'm getting exactly the same. Odds of getting a working number, are like the
odds of winning the lottery.

My guess is they are either trying to find a voip trunk, or they are trying
to make cold calls to the extensions on my system. Sales or something
similar.

 

We got pounded last weekend, but installed a list of distant IPs in IPTABLES
and see nothing this weekend.

We have no need to be contacted by any sites more than 2500 miles away, and
not too many from within 2500 miles. ;-)

Cary Fitch

 

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Re: [asterisk-users] Why are the hackers scanning for these?

[Cary Fitch]

My guess is they are looking for 10 digit phone numbers as extensions.

 

Are they all from 1 IP address or from many?  If from many, they are likely 
many serial scan or from a list of suspected VOIP numbers.  If from one, and 
that random, then from a list of suspected VOIP numbers.

 

Since you listed a phone number as part of your signature… I might guess 
hackers might soon add that number to a scan list.

 

It is one thing to randomly run 2,XXX-, to 999-999-, with skips for the 
“dead zones,” (0-XXX-XXX-) etc. but another to hit suspected VOIP numbers.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Murphy
Sent: Sunday, November 07, 2010 8:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Why are the hackers scanning for these?

 


Hey, I'm going thru logs, and I see some very common and interesting things
that the hackers are looking for.

In a whole bunch of scans, I've noticed that the first guess or two for sip 
accounts
is usually a 10-digit number. I'm asking myself, why these numbers? Are they 
looking
for a voip trunk? Or is it just like a serial number for the scan? What?

Here's some examples:

2648061411
3190339404
2685608247
3358171034
2092652562
2206598858

Just trying to follow the advice: "Know thy Enemy"

murf



Steve Murphy

ParseTree Corp.

57 Lane 17

Cody, WY 82414

✉  m...@parsetree.com

☎ 307-899-5535

 
<http://www.wisestamp.com/email-install?utm_source=extension&utm_medium=email&utm_campaign=footer>
 Signature powered by  
<http://www.wisestamp.com/email-install?utm_source=extension&utm_medium=email&utm_campaign=footer>
 WiseStamp 

  
<http://s.wisestamp.com/pixel.png?p=mozilla&v=2.0.3&t=1289138760949&u=949715&e=4286>
 

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Re: [asterisk-users] Phones slow to ring

[Cary Fitch]

Watch the console as you dial.  Dial the number and "#".  The ring should be
"instant".  Or if not, look at the console and report what you see.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jy
Sent: Thursday, November 04, 2010 5:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Phones slow to ring

 

I am new to asterisk and using it for a research project.  Have set up an
server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are
registering fine with the server.  They are able to call one another,
however, the problem is it takes roughly 8-10 seconds for the called phone
to ring.  I have a really simple dialplan using only 4 digit extensions and
have turned off callerid. Both phones are on the same subnet and I have
enabled nat and keepalive.

Does anyone have an idea what could be wrong here or idea on how to debug
this problem?

Thanks,
John

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Re: [asterisk-users] FW: Under heavy attack

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Tuesday, November 02, 2010 10:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FW: Under heavy attack


>
> I'm still on old-fashion copper-wire and "have yet to experience the joy"
of
> SIP Trunk-"ing" and the type of issues discussed in this thread.  My
thought
> to share here is that outgoing calls should be "easy" for thoroughly
> authenticated users and impossible for others...
>
> Probably more can-o-worms than help.  Sorry if this is so.
>
>
>

nothing new here, this is just the digital equivalent of a wats line 
with a weak access code for outbound access.
the difference is code guessing can be a lot more aggressive now, and 
finding the inbound path is simpler.

==

Each system needs to be configured according to its purpose and needs.
Simply these are phone systems, not e-mail or web servers.  You may want to
be able to get mail from (almost) anywhere in the world, same for web
services.

But for a phone system you may have very different needs.  One can visualize
the differences between a national or international VOIP provider, a 4
person office in Little Rock, AR, a local SIP provider in Houston, TX and an
international sales company with offices in Rome Italy.

A small sip system used with an upstream VOIP provider should be invisible
to 99.% of the world's population. (Excepting any other trusted peers.)

If there was a wide spread peering network and an individual system
wanted/needed to access and be accessed like email then it would be a
different world.  We could all be robo-call spammed just like email. :-(

But leaving small systems open for attack from 99. percent of the world
is just begging for trouble.

Cary Fitch


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Re: [asterisk-users] FW: Under heavy attack

[Cary Fitch]

 

 

I know there was talk on VUC recently about some kind of realtime RBL for
SIP. Has anything progressed?

 

It would be SO easy for asterisk users to contribute to a blacklist and also
do a lookup in realtime to see if an IP has been blacklisted.

 A little bit of joined up thinking in the community could eliminate this
issue. Would also be another major + for Asterisk as a platform..

 

Regards

Brian

 

 

 

Some systems need to communicate with the world.

 

Other only with their own network, and a few selected outside addresses.

 

If anyone from Amsterdam or Nigeria or Malaysia (and 100 other countries) is
trying to get on our system, we are surprised!  Vail Colorado, not so much.
:-)

 

Cary Fitch

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Re: [asterisk-users] FW: Under heavy attack

[Cary Fitch]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256

Here's my take on the attack... Sigh...


http://www.stuartsheldon.org/blog/2010/11/sip-brute-force-attacks-escalate-o
ver-halloween-weekend/

Stu


They were trolling for SIP account IDs, not really trying to register.
It was a coordinated bot or spoofed source attack not "The Halloween Club"
doing tricks.

Any small system should:

Use IPTABLES and block any parts of the world you don't need access to/from.
Start with any Class A address that is probing your system.

Make your SIP IDs 8-12 characters in length, and use at least alpha &
numerical characters, some special characters if you like a little more
variety.

 bear3579
 b3e5a7r9
 Bear3579
 La3579ke

Or more.

Do the same for passwords.

6543office
7659home

Etc.

Are these perfect?  No, but they are human friendly, and require the
exploiter to hack a 16 to 24 character combination ID and Password that has
36 or more characters in the character set.  Of course some dashes or
periods or commas or others can be added.  And when you see an attack if it
isn't from a network on your planet, put the whole network in IPTABLES.

(And get the world country delegations for IP addresses and block all "not
on your planet.)

$.02
Cary Fitch


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Re: [asterisk-users] FW: Under heavy attack

[Cary Fitch]

I was going to point out a failing of the attackers, but figured they read
the list and don't need any more tips.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Monday, November 01, 2010 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: Under heavy attack

 

And obviously these attackers read our emails on lists like this and adjust
their sick strategies accordingly.

Zeeshan A Zakaria

--
www.ilovetovoip.com
www.pbxforall.com (beta)

On 2010-11-01 12:02 PM, "Jamie A. Stapleton"
 wrote:

Only 100?  We had a single server over 300.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Saturday, October 30, 2010 9:49 PM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Under heavy attack



 

My count has reached 100 for the day. The server serves doesn't serve
international calls anywa...

Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak  wrote:

No.  It seems that opening ...


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Re: [asterisk-users] Under heavy attack

[Cary Fitch]

We have about 8-10 boinking us.  They generally run a 1- peer attack and
a few alphas like common words or "eieio"  We use large, complex peer IDs
and passwords, so they have a long way to go.   I am happy to help keep them
busy.

I also send messages to their network abuse address.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Saturday, October 30, 2010 6:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Under heavy attack

On Sat, 2010-10-30 at 14:28 -0400, Zeeshan Zakaria wrote:
> My main asterisk server is under unusual heavy attack, and so far
> Fail2Ban has blocked about 30 IPs, from various different countries.
> At this time it is blocking about 1 IP address every few minutes.
> 
> Just wondering if anybody else is also experiencing unusually
> increased hack attempts today?
> 

Just 30 ?

I got 1593 different IP's on my personal blacklist who constantly are
looking if i may lower my guards. Though 82.101.63.5 and 132.68.58.60
are rather busy tonight...

hw

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Re: [asterisk-users] ISDN & SS7

[Cary Fitch]

I do not have knowledge of the SS7 vendors for Asterisk.  Using redundant
56k data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN
on a commercial telephone switch, with no issues at all.

 

SS7 can support any number of simultaneous calls depending only on the
bandwidth of the SS7 channels.  SS7 is always done on a redundant channel
basis since it is so important.  

 

Cary

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN & SS7

 

Hi cary,

 

Can you recommend me what add-on vendors I should use ?

Can a open source solution such as chan_ss7 or libss7 support many
conncurrent calls (for example 240 calls) ?

 

Thanks

 

  _  

From: Cary Fitch 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Sent: Sun, October 24, 2010 9:33:28 AM
Subject: Re: [asterisk-users] ISDN & SS7

SS7 is an inter-telco system using a separate network for all signaling.

 

You must have an SS7 network connection before anything will work.

 

Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
data and connection info between the switches.

 

Asterisk doesn't support SS7 natively although I believe there are one or
more add-on vendors.

 

Cary Fitch

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7

 

Hi all,

 

I'm being requested to deploy an IVR service using SS7. 

I've deployed Asterisk before using ISDN connection, but never with SS7.

Can anyone explain me the different between using ISDN and SS7 ? What need I
do now to change to use SS7 ?.

 

Many thanks,

Giang

 

 

 

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Re: [asterisk-users] ISDN & SS7

[Cary Fitch]

SS7 is an inter-telco system using a separate network for all signaling.

 

You must have an SS7 network connection before anything will work.

 

Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
data and connection info between the switches.

 

Asterisk doesn't support SS7 natively although I believe there are one or
more add-on vendors.

 

Cary Fitch

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7

 

Hi all,

 

I'm being requested to deploy an IVR service using SS7. 

I've deployed Asterisk before using ISDN connection, but never with SS7.

Can anyone explain me the different between using ISDN and SS7 ? What need I
do now to change to use SS7 ?.

 

Many thanks,

Giang

 

 

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Re: [asterisk-users] dials a trunk when off hook

[Cary Fitch]

I am not sure that can be done literally by Asterisk because most
phones/adapters give dial tone when off hook, but Asterisk doesn't know the
phone is off hook until a send button is pushed, several seconds pass after
some keys are pressed, or the # button is pressed.

However many of the adapters can be set to autodial.  I would look for a
phone or adapter that has autodial ability.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH
Sent: Friday, October 22, 2010 6:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] dials a trunk when off hook

How can I let asterisk immediately dials a trunk when off hook?


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Re: [asterisk-users] SIP Blacklisting

[Cary Fitch]

We would be interested.

Spam is a harder problem to fight due to volume and the ability of any idiot
to set up free email accounts. But anyone blasting SIP systems is a pure
commercial crook. Tagging and strangling them should be a clear cut project.

Cary Fitch


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, October 21, 2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Blacklisting

Hi,

Given the recent increase in SIP brute force attacks, I've had a little
idea.

The standard scripts that block after X attempts work well to prevent you
actually being compromised, but once you've been 'found' then the attempts
seem to keep coming for quite some time. Older versions of sipvicious don't
appear to stop once you start sending un-reachables (or straight drops). Now
this isn't a problem for Asterisk, but it does add up in (noticeable)
bandwidth costs - and for people running on lower bandwidth connections. The
tool to crash sipvicious can help this, but very few attackers seem to obey
it..

The only way I can see to alleviate this, is to blacklist hows *before* they
attack. This means you wont ever be targeted past an initial scan.

Is there any interest in a 'shared' blacklist (similar to spam blacklists,
but obviously implemented in a way that is more usable with
Asterisk/iptables)?. Clearly it raises issues about false positives etc, but
requiring reports from more than X hosts should alleviate this. There's all
the usual de-listing / false-listing worries as with any blacklist, but the
SMTP world has solutions we could learn from.

Leaving a 'honeypot' running on a single IP address has revealed a few
hundred addresses in less than a month. I am fairly certain these are all
'bad' as this host isn't used for anything else. There is obviously a wealth
of data (and attacks) out there that would be good to share.

Anyone have any thoughts?

S
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Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician

[Cary Fitch]

His post may have been of interest to some outside of DFW, and I appreciated
your post less than his.  But, enjoy.

C
==

A very vast majority of people on here are not in Dallas (and indeed
probably a majority in the US). So stop filling their mailboxes with this
crap.

Incase you hadn't noticed "Asterisk Users Mailing List - Non-Commercial
Discussion"



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Re: [asterisk-users] fraud advice

[Cary Fitch]

As a practical matter, on anything that can generate endless billings, there
should be a dumb trap that compares current usage to history (last month)
and if usage exceeds 2/1 or 3/1 for instance then usage is choked or denied
enough to cause the user to complain or perhaps generate a message to call
customer support, (or call your cell phone!)

Then if it is valid, raise last month's reference enough to let current
calling continue.  If it isn't valid you have found a problem and saved your
or your customer's caboose.

As to who to complain to, gather all info possible and report to everyone
you can find.  Someone may investigate, but there isn't likely anyone who
will absolve the problem.  Some will just take the report and ... as far as
you are concerned, do nothing.  There isn't much a local police dept. can do
about a hacker in Western Slobovia cracking your server.

Generally the FBI doesn't take matters of less than $10,000.  But it sounds
like you may meet that test.

But they could take months or years or never finding the culprit and finding
the culprit will likely net you nothing financial for you will be 1/10,000
of the fraud they did.

This is a problem like spam in email.  But this has cash costs to the server
operator/customer.  Passwords need to be un-crack-able, and there should be
usage alarms, as described above.

Depending on the situation even a single counter to your upstream billable
sip server for all usage would likely trip on excessive usage and save your
bacon. 


Cary Fitch





-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, October 14, 2010 8:11 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] fraud advice


Hi,

Embarrassed as I am to write this, I am hoping for some advice.  One of 
our very first PBX installs, now six years old, was "taken advantage of" 
over the past few weeks.  A victim of sipvicious, I assume, that managed 
to guess one of the SIP passwords.  4000 calls to various middle eastern 
destinations have been placed, which ended up being sent over our 
customer's PSTN trunk, and of course there was no warning until the bill 
came today.  Unfortunately the bill only covered the first few days of 
this fiasco, and was only $700.  I am afraid the one that is on the way 
will be tens of thousands.  ONE CALL on the bill that just arrived was 
$200 (80 minutes to Sierra Leone).

I'm sure this started out as a single scan.  It must have been posted, 
because I have at least ten IP addresses now that were placing calls via 
the same peer.  They are from all over the world.

So what is the accepted procedure?  I'm in the US Virgin Islands, so do I 
go to the FBI?  Police?  Is their some telecom fraud body to report such 
things to?  Does any one ever get any relief from such events?

I'm basically sick to my stomach right now.

j

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Re: [asterisk-users] Big time system

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Elvish
Sent: Saturday, June 26, 2010 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big time system

Hi All,

I am not sure if my comments will be helpful but here goes.



Let me know if you need anymore pointers. Also happy to consult but
you would need to contact me off list for that...

Stuart Elvish

===

Thanks, your info is most helpful, as is the other info I have received,
some of it in private messages.

Your snapshot description of a ~4000 user system and architecture is a good
starting point for our planning.

Cary Fitch


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Re: [asterisk-users] Big time system

[Cary Fitch]

Thanks for the feed back, but the rates are more or less predetermined.

AT&T rates would be $.0007 per minute for local calls.  The operation would
be providing local phones wired to houses with copper pairs.

What I am looking for is the "best" ways to handle those lines when brought
to a local "switch" site.  The actual "switch" might not be there but back
hauled, might be a TDM switch, a concentrator (TNT, etc) "10" ganged
Asterisk systems, or "tin can and string". 

I see some talking about TNTs in this forum.  Those are 672 lines or in some
versions double that, what is used behind them to do the processing, etc.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 25, 2010 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big time system

On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch  wrote:
> But, we have an opportunity to get into a big time telecom activity.
>
> It would have 2000 to 30,000 user lines per city, and we would like to
have
> those brought back to a central location for control and because transport
> can be more economical than remote site rentals, maintenance and
personnel.

I would say you need to make an RFP process to first negotiate your
calling rate extremely low with the major vendors of the country where
you're operating. If this is US, you're talking Qwest, AT&T, Verizon,
and the ilk, and you negotiate an extremely low minute rate in return
for giving them a guaranteed minimum revenue. And while you're at it,
you ask them how they suggest you design the architecture over their
national network.

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[asterisk-users] Big time system

[Cary Fitch]

We are an asterisk user... small time system 50-100 users or so.  

But, we have an opportunity to get into a big time telecom activity.

It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can be more economical than remote site rentals, maintenance and personnel.

We could take the local lines into concentrators (TNTs or equivalent) and
bring back IP to a central site, or put servers at the remote cities.

Our object is to serve as a "central office switch" for subscribers on
standard telco service loops.

This isn't a "How many lines can I handle using a Belchfire 2600 processor?"
type question but a request for pointers to big time systems.  There would
be no IP path to the end user, "just" copper.

Thank you
Cary Fitch


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Re: [asterisk-users] Why asterisk down when inet server down?

[Cary Fitch]

But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No 
sip. No iax. Why does the asterisk machine have to resolve any address?

The internal phones can't even call each other, even though they have 
hard ip addresses.

> Same for doing DHCP for handing out addresses to your phones...
>

All the phones have manual ip addresses. No DHCP.

sean

Do the phones find the sip server by IP or by domain name.

I.e.   1.1.1.1
Or sip.yourdomain.com

If domain name, what are they using for DNS?

Cary Fitch


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Re: [asterisk-users] "ring splash"

[Cary Fitch]

The ring splash is a long standing feature of call forwarding.

Of course somewhere in the Asterisk code a change could be made to extend
the time required to detect a valid ring.

But, how about just unplugging the pots lines from the PBX with a quick
restore ability?  Unplug lines at the NID, or open bridging clips or
whatever applies.

Cary Fitch



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, May 26, 2010 11:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] "ring splash"


Something new to me.  Recently installed a 1.4.30 box for a small office 
with four POTS lines in a hunt (Digium TDM410P).  Had the telco put a 
"call forward" option on the main line of the hunt.  They dial a feature 
code from their desk phones (Polycom IP450) that results in forwarding the 
main number to our VoIP service.  This is all to let them "try out" our 
dialtone service before porting the number to us and ditching the POTS 
lines.

So we perform some test calls and they all go through fine, and everyone 
is happy, BUT everytime a call comes through it ALSO causes the POTS line 
to ring, and a "ghost" call rings all the phones in the office (the 
desired result of an inbound call from POTS).  When they answer it they 
get fast busy because it isn't actually a real call.

I spoke to the telco this morning about it and they said "oh yeah - that 
is a "ring splash" that lets the customer know that a call was forwarded". 
They said this was a feature of their DMS-100, it has worked that way for 
twenty years, and they can't turn it off.

So to the question - can the TDM410P somehow tell the difference between a 
"ring splash" and an actual inbound call?  I think in the meantime I will 
send inbound POTS calls to an auto attendant that will eventually hang up, 
but would love a more elegant solution ;)

Cheers,

j

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Re: [asterisk-users] NPA NXX Database

[Cary Fitch]

http://www.localcallingguide.com/

 

will give you lots of info.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, May 18, 2010 12:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] NPA NXX Database

 

Has anyone had good results with an on-line database that returns a LATA
based on NPA NXX?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

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Re: [asterisk-users] MATH

[Cary Fitch]

Clue, 

If a caller keys in "4 5 3" will some variable return "453"?

I ASSume yes, since you can make menu selections with DTMF, "obviously" you
can process the results further or in other ways than that.

Cary




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Sunday, January 31, 2010 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MATH

On Sun, 31 Jan 2010, Thomas Perron wrote:

> does dtmf any any variable that i can capture and use w/ some logic
> like in the case of a gotoif

Anyone have a clue what this means? Anyone? Anyone?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] disable comfort noise

[Cary Fitch]

Something else that is flaky, missing or otherwise irritatingly broken
in the piece of shit that is 'Asterisk'.

[Cary Fitch] 
It is an open source project.   When can we count on your contribution of a
comfort noise generator that will not be a "piece of s--t?

Can you have that by Monday?

CF


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Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem

[Cary Fitch]

As a guess, they can both talk to the server, but can't talk to each other.


What is common to that is they may be trying to reinvite each other, and
there is no path through the respective routers/firewalls to the other.

So if reinvite is set to yes, set it to no, in both phone profiles on the
server.

Cary Fitch



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu
Sent: Monday, January 25, 2010 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip.conf with versatel and two NICs very
strangeproblem

Hi

My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has 
the local ip 10.26.208.252
and the external ip 89.244.x.y

eth0 of the server is configured to 10.26.192.107

The Problem:
SIP registration works, phone rings in- and outbound, but there is no 
audio, nor the caller neither the callee
can hear anything.
So i am quite sure that is has something to do with firewalls, natting 
and so on but i?ve read hundreds of
pages and tried thousands of setting but i cant get audio to work..
the strange thing is... when i call the versatel-sip-number from my 
mobile phone, i see the call coming in
in the cli, i see the voiceprompts that asterisk plays, but even there I 
cant hear anything on my mobile.
next strange thing:
i defined 2 sip-extensions. both are registered... everything is fine... 
routes are ok, they can call out
and can be called from external and from internal (sip phones call each 
other).. but the same... no audio.
but when one sip extension calls a wrong number... the "cannot be 
completed" message is hearable.
i configured a queue with moh and even this works... but why cant to 
sip-phones talk to each other?
why cant an external caller hear any audio?

if i make sip debug, i see traffic (and due to extension is calling i 
think that on the sip-level everything
is okay...) how can i see, which port and interface is chosen for audio 
when a call comes in?

thanks,
yves


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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[Cary Fitch]

If the phone is first, then it slightly limits the PC and rebooting the
phone causes loss of contact with the PC.

If the PC is first you have to have dual ports on it (a few bucks of
hardware, plus configuration costs), then rebooting the PC causes the phone
to loose contact with the world.  Not good if you are on the phone and need
to reboot.

Two separate feeds would work best, but cost more.  Dual wall jacks with
"green-PC" and "blue-Phone" jacks could then be used.

The phone will be on the desk, the PC may be under it.  Two jacks, two
cables.  

If there were a standard for two Ethernet connections in a cable... that
could work, but might interfere with Power Over Ethernet.

I wouldn't want to be like Bill Gates saying "640K memory is enough for
anyone circa-197?", but isn't two 100 meg connections enough for any single
desk?  The phone doesn't really need more than 10 meg.

An advantage of the separate net for the phone, is that it would make POE
easy for the phones, and eliminate a lot of wall warts under every desk.
Plug in the phone and it works.

YMMV.

Cary Fitch


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Re: [asterisk-users] SIP over VPN -- no audio to other remote/VPNconnected phones

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack
Sent: Monday, January 11, 2010 1:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP over VPN -- no audio to other
remote/VPNconnected phones

Hello,

I am having a problem with my current SIP over VPN setup.

We have a server running asterisk at our office.  All the phones in the 
office are on the same network / local to this server.  We also have two 
employees with home offices using SIP phones over VPN to connect to the 
asterisk server.  These phones have no problem with calls to the phones 
in the office, however there is no audio when trying to place a call 
from one remote phone to the other.

I'm assuming this is a routing issue, but I have no been able to come up 
with a solution.

Thanks for any advice.



[Cary Fitch] 

One thought: if you are using "reinvite" try turning that off. That will be
a clue.

It would seem that both phones are on the local net via VPN, and should be
able to talk to each other if they can talk to anyone in the office. (As you
know.) So look for clues as to real paths to each other.

Cary Fitch

 

 


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Re: [asterisk-users] Please remove me from the mailing list.

[Cary Fitch]

I have been doing "this" (whatever that is), since about 1976, involving
many facets, including posting on #1 "CBBS" out of Illinois, usenet in the
90s, and more.

It is not possible to get people to follow all the RFC "rules and customs"
much less the "-- CR" sigdashes.

There are a lot of relative "newbies" much less "oldies" who never heard of
such, or run 10 different mail options from gmail to hotmail to sendmail to
"I have no idea what it is, I am just a member, poster, customer,
or-something " mail user.

In the 90s, a very well like member of a BBS type system
(MajorBBS/Worldgroup) went ballistic when people started using HTML.  The
other people on the net finally just told him "We don't care, we are not
staying in the dark ages. Like it or lump it.".

I am on numerous lists where 75% to 100% of the posts are top posts.

If someone bottom posts people who want to go to the bottom and read.  Most
of us start at the earliest post and read message by message.. and don't
want to rescroll through 10-20 messages over and over.  I know that it would
be nice to have the last message have all the text inline, but that doesn't
happen either.  And then there are always 5 more messages in the same thread
later today.

I don't even use a sig file.  I just type my name.

But to see if it works:

-- 
Cary Fitch


   IMHO, top-posting isn't the problem, but just an obvious symptom of the 
real problem, which is failure to edit/strip the quotes to the bare 
minimum.  When a thread gets hijacked by top-posters, who bang out their 
thoughts without even scrolling down to see all the garbage below, another 
problem also becomes apparent, and that is the failure of many MUAs to 
honor 'sigdashes', which is the convention of preceeding your sigfile with 
a line that is 'dash dash space '.  A compliant MUA will strip that 
line and everything after it when quoting for a reply or forward.  Note 
for the list admin:  Please preceed your message-footer with a sigdashes 
line!

-- 
Rick Green

"Those who would give up essential Liberty, to purchase a little
temporary Safety, deserve neither Liberty nor Safety."
   -Benjamin Franklin

"As for our common defense, we reject as false the choice between our
safety and our ideals."
-President Barack Obama 20 Jan 2009

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Re: [asterisk-users] Dialplans & Holiday Dates

[Cary Fitch]

Perhaps make the dates a database entry?  The fixed dates would stay the
same each year and you would adjust only the floating dates.

Or, there are really few holidays in the year. (Unless you are a government
or a bank)  Simple intercept code in the dialplan would handle most
businesses.  

"Just" write 5-10 cloned lines of date traps in the code to pass the calls
or send them to a "closed" handler.  That is less disk/system intensive that
doing a disk access, except they would likely be in cache anyway.

Cary Fitch



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Thursday, December 31, 2009 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialplans & Holiday Dates

I have a working dialplan for our phone system with Mon-Fri, business 
hours identification, etc.  But what I'm lacking right now is support 
for company holiday dates.

What I'd like to do is to create a database of these dates and just 
update them as new years rollover.

I suspect others have done this sort of thing with Asterisk before, but 
I've not found any resources so far.

Does anyone have a suggestion as to how to approach this?  I'm running 
Asterisk 1.4.2.

Thanks
Myles
-- 
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona  USA
http://www.techsolusa.com
Phone +1-480-451-7440


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Re: [asterisk-users] Can't get G.729 to work...

[Cary Fitch]

And tell Asterisk that G.729 is the only codec for that number as well!

Cary Fitch
 



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Re: [asterisk-users] Can't get G.729 to work...

[Cary Fitch]

Watch the calls on the console.  Try both ways. Document what you see and
your codec settings on both the phone, and sip.conf.

You may have to tell the phone that the only codec it can use is G.729,
don't just make that "first choice". Make it the only choice.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...

I thought I already did that - which is how they now get some (but not
yet all) of their calls on G.729.  

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr & Tower
www.rolandschorr.com
b...@rolandschorr.com


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Re: [asterisk-users] FW: hi Dan

[Cary Fitch]

My point was the two previous posters could have ignored the request and
made no post at all.  That they were "violating" a "rule" by top posting to
tell a person not to bug them.

And, someone criticized me for an off topic post and of course there have
been 15-20 more.  And some have top posted and interleave posted, and etc.

And, it will all die down in a day or so.

It is Friday night, time to turn off the computer and click "Mark all as
read" on Monday morning. 

Be charitable and kind.  Have a nice weekend.

Cary Fitch



 

  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, November 13, 2009 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: hi Dan

On Fri, 13 Nov 2009, Cary Fitch wrote:

> Sorry, I can't resist.

Next time please try harder.

> How do I join the Mail List Nazi Corp?

And of course, lacking any sense of "history," let's blame it on the 
Nazis.

> Asking neophyte questions are objected to by some, top posting by those 
> who blast others, etc.

Not at all. If you had any sense of Pawan's "history" you may have chosen 
sides differently:

Date: Tue, 27 Oct 2009 09:34:18 +
Subject: [asterisk-users] Installing Asterisk

Pawan states he is reading an Asterisk book and requests suggestions on 
which OS he should use.

He received helpful responses from Dan Journo, PATRICK KANGETHE, John 
Novack, and Hans Witvliet.

30 minutes later he posts a brilliant tome "Subjected" "installing" 
consisting of 2 words -- installing asterisk. He received less than 
helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno.

Date: Wed, 28 Oct 2009 14:07:30 +
Subject: [asterisk-users] deploying asterisk

Pawan states he "had just finished the installation requirement of 
asterisk" and now feels competent to piss off 40 executives with his first 
installation.

He received helpful responses from Danny Nicholas, Darrick Hartman, Steve 
Edwards (me), and Alex Balashov.

Date: Mon, 02 Nov 2009 09:37:42 +
Subject: [asterisk-users] hardware requirements for asterisk

Pawan request help with hardware requirements. Curiously, he implies that 
he can read and has "just finished my chapters of asterisk."

He receives helpful responses from Alex Balashov and Hans Witvliet.

Date: Fri, 06 Nov 2009 04:33:09 +
Subject: [asterisk-users] asterisk,libpri,zaptel

Pawan requests help installing Asterisk.

Date: Fri, 06 Nov 2009 17:08:02 +
Subject: [asterisk-users] problem while compiling asterisk tar file

Pawan requests help in compiling gtk.

He receives helpful responses from Jimmy Godbout, Danny Nicholas, Steve 
Howes, Jason Parker.

Date: Sat, 07 Nov 2009 17:29:57 +
Subject: [asterisk-users] help in installing asterisk

Pawan requests help in compiling Asterisk.

Date: Sun, 08 Nov 2009 06:20:46 +
Subject: [asterisk-users] how to check version of asterisk

Pawan requests help to determine the version of Asterisk he installed.

He receives helpful responses from Alex Balashov, Tzafrir Cohen, and C. 
Savinovich.

Date: Mon, 09 Nov 2009 17:11:47 +
Subject: [asterisk-users] how to configure softphones in asterisk

Pawan requests help configuring a softphone. He does not indicate that he 
has done any research, tried anything or received any error messages.

He receives helpful responses from Matt Riddell and Danny Nicholas. He 
receives less that helpful responses from Alex Balashov, Steve Howes, and 
C. Savinovich in response to emailing them privately.

Date: Tue, 10 Nov 2009 18:16:50 +
Subject: [asterisk-users] how to configure softphones in asterisk

Pawan solicits help configuring a softphone. He does not indicate that he 
has done any research, tried anything or received any error messages.

He receives helpful responses from Alex Balashov and Barry L. Kline.

Date: Thu, 12 Nov 2009 06:31:35 +
Subject: [asterisk-users] soft phone (X-lite) not able to register
with asterisk

Pawan solicits help configuring a softphone. 20 minutes later he posts the 
same request.

He receives a helpful response from ABBAS SHAKEEL.

Date: Fri, 13 Nov 2009 08:47:41 +
Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE

Pawan indicates he has succeeded in placing a call between 2 extensions 
and now wants someone to complete is dialplan.

He receives a helpful response from Leif Neland and less than helpful 
responses from Steve Howes and Steve Edwards. He also invites a flame-fest 
by soliciting help privately from several list members.

All this in the last 2 weeks.

> Some people have no one within 250 miles of where they are to learn from 
> or learn better by working with code than reading inscrutable examples 
> from different versions, and other inanimate pages of examples th

Re: [asterisk-users] FW: hi Dan

[Cary Fitch]

Slightly paraphrasing a very old and wise saying:
Give a man a fish,
he eats for a day.
Teach him how to fish,
he eats for a lifetime.
-- 
JohnM

I see no teaching, just no help.

He doesn't eat today or tomorrow either.

Cary Fitch

 




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Re: [asterisk-users] FW: hi Dan

[Cary Fitch]

Sorry, I can't resist.  

How do I join the Mail List Nazi Corp?  Do I have to be invited, or can I
just self appoint myself?  Asking neophyte questions are objected to by
some, top posting by those who blast others, etc.  

How about leaving member chastisement to the sponsor of the list?

Some people have no one within 250 miles of where they are to learn from or
learn better by working with code than reading inscrutable examples from
different versions, and other inanimate pages of examples that have "wrong"
variables, etc.  

Nearly everyone can be criticized for something, Asking "dumb" questions,
top posting, bottom posting and leaving 3 pages of "crap" to scroll through,
answering questions that were answered 5 posts down, because they didn't
review the newer messages before posting, and more.

Be charitable and kind.  Have a nice day.  

Cary Fitch


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, November 13, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: hi Dan

Its not just you mate. He's doing it to everyone, and sadly the list  
server is too clever to accept forged unsubscribes..

Steve

On 13 Nov 2009, at 15:22, Dan Journo wrote:

> Please stop emailing me personally.
> If no one replies to a post, it means that everyone is busy or they  
> think you should read through the documentation before posting.
>
> If you can't figure out simple things like Music on hold from the  
> documentation, then i dont think VOIP is for you.
>
> -Original Message-
> From: aster...@opensourcesolution.in
[mailto:aster...@opensourcesolution.in 
> ]
> Sent: 13 November 2009 09:18
> To: Dan Journo
> Subject: hi Dan
>
> Hi dan,
>
> sorry for sending u personal mail. i am a beginner in asterisk, i  
> had configured a minimum dial plan in which i had made two  
> extentions n made call between two extentions via soft phone (X- 
> Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz  
> if u have any good link or documentation than share it with me.
>
> Regards,
>
> Pawan
>
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[asterisk-users] "POTS 4K linear codec"

[Cary Fitch]

I am not sure what the problems are and the reasons for the basic 64K modems
used in VOIP are.  I understand the compressed codecs that get the bandwidth
down to 20-30 K.  And perhaps the 64K units give much better potential audio
than you would get on a normal POTS line.

But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old
phones.

Multiple transcodings cause issues.  Today a cell phone or a POTS line phone
can send DTMF clearly enough to operate a credit card or other interactive
tone based system at the far end.  With SIP it is sometimes "chancy".

Is there a plain 64K codec that would simply pass through the SIP server and
be handed off to a PRI or phone co. trunk on a T1 on the other side of the
SIP server?  Digital 64K telco sounds very good as a phone conversation.

Cary Fitch





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Re: [asterisk-users] OT - Number Portability

[Cary Fitch]

Two more comments.

Yes, to join the PSTN call distribution system you must have SS7.

While rural ILECs are not exempt from number portability, there is a court
injunction that saves them from having to transport the call out of their
local rate center, so getting calls from a distant RILEC to a central point
is at a cost to the requesting carrier.   There are other complexities.

Cary Fitch



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Saturday, October 31, 2009 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT - Number Portability


Sorry for the off-topic, but perhaps this will be of interest to other 
asterisk based ITSPs.

We are starting service in a rural area where the ILEC has the rural 
"monopoly".  From what we have read in the FCC docs this does NOT exempt 
them from number portability, but what does it take for us to qualify to 
receive their numbers?  To date we simply have a few voice trunks to them, 
and a set of DID numbers we purchase from them.  Do we have to be a full 
CLEC to participate as a carrier?  Does this imply we must have an SS7 
connection to the PSTN?

Thanks for any info,

j

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Re: [asterisk-users] OT - Number Portability

[Cary Fitch]

Your chances are likely slim to none.  But good luck.

First to port numbers you have to be a recognized carrier, which for the
most part means getting numbers from NANPA : North American Numbering Plan
Administration.  To do that you have to be certified by your state PUC or be
a CMRS (cell phone) carrier.

They would give you a block of 10,000 numbers designated to the rate center
of the ILEC in question.

Then you designate on of those numbers as a "local routing number" (LRN)
which is like a "pathfinder" number for ported numbers.

And, you work out an Interconnection agreement with the local Telco
(probably with them kicking and screaming for months or a year) because they
really don't want you there, and you aren't a big cell phone company, but a
local wire line competitor, which then is approved by the state PUC.

What some others have done is to operate as a "PBX Service provider" or some
other business term.

They get a PRI from the local company, and become the agent for the
customer, move the service delivery to their PRI, and then distribute the
calls to the appropriate customer via SIP and Asterisk or other solution.

That has worked in Casa Grande, AZ for one place.

(Not ours.)

Cary Fitch





-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Saturday, October 31, 2009 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT - Number Portability


Sorry for the off-topic, but perhaps this will be of interest to other 
asterisk based ITSPs.

We are starting service in a rural area where the ILEC has the rural 
"monopoly".  From what we have read in the FCC docs this does NOT exempt 
them from number portability, but what does it take for us to qualify to 
receive their numbers?  To date we simply have a few voice trunks to them, 
and a set of DID numbers we purchase from them.  Do we have to be a full 
CLEC to participate as a carrier?  Does this imply we must have an SS7 
connection to the PSTN?

Thanks for any info,

j

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[asterisk-users] Grandstream 2010

[Cary Fitch]

The Grandstream 286s automatically re register when a connection is
restored.

Our Grandstream 2010s don't.  Does anyone know of a setting that makes them
reregister?  I has tweaked "Watchdog timer" and anything that looked
promising.

Cary Fitch
Affordable Telecom


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Re: [asterisk-users] Can dial long distance but not local?

[Cary Fitch]

Perhaps send it as 10 digits or 1+? 

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Wednesday, October 07, 2009 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can dial long distance but not local?

 

AsteriskNOW 1.4.26.2 with a  Digium TE205P connected to an ISDN PRI (single
span).  I'm sure I just have something goofed up in the dialplans?  I have a
bunch of Polycom 331 IP phones connecting to the server.  I can dial the
other extensions in the system fine and I can dial long distance outgoing
but cannot seem to get it to dial local (7 digit) calls.

 

I see this in the CLI:

 

-- Executing [...@macro-dialout-trunk:19] Dial("SIP/801-09b6e498",
"DAHDI/g1/5551212|300|") in new stack 
-- Requested transfer capability: 0x00 - SPEECH 
-- Called g1/5551212 
-- Channel 0/1, span 1 got hangup, cause 28 
-- Hungup 'DAHDI/1-1' 
== Everyone is busy/congested at this time (1:0/0/1) 
-- Executing [...@macro-dialout-trunk:20] Goto("SIP/801-09b6e498",
"s-CHANUNAVAIL|1") in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) 
-- Executing [s-chanunav...@macro-dialout-trunk:1]
GotoIf("SIP/801-09b6e498", "1?noreport") in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) 
-- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp("SIP/801-09b6e498",
"TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 2Cool - failing through
to other trunks") in new stack 

Also when I check the PRI DEBUG I see an Error 28 which indicates an invalid
number format.  But I'm just sending 5551212, which should be o.k.

 

I'm a newbie at this.any suggestions welcomed.

 

-Ben-

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Re: [asterisk-users] New thread - SIP over VPN

[Cary Fitch]

Last week I did a Microsoft VPM from one XP computer to another via Verizon
broadband wireless.

SIP worked ok, but BLF on a Grand Stream 2010 didn't work. 

In addition to the VPN the phone was behind a NAT router.  The phone was
already set up behind the NAT Router, the only difference was to get the
connectivity via Wireless VPN.  There could have been some missing ports in
the VPN environment.

The audio was good, but there were times it lost clarity, likely to wireless
bandwidth/lag/jitter issues.  I decided that couldn't be my main business
phone.

Cary Fitch


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[asterisk-users] Voiced E-mail

[Cary Fitch]

Does anyone have info or starter points on how to take emails from an
external POP3 or IMAP server and cause them to be voiced by Asterisk?

It is our e-mail server, so we can do anything to it.  My question is
concept or products required to get asterisk to do the job.  Text-to-voice
converter? Program to strip email down to just to, from, text, special mail
box, have it call user, or have user call in?  Whatever anyone that has done
something like this would suggest.

Thank you.

Cary Fitch


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Re: [asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657

[Cary Fitch]

FWIW:
>From "old, old" memory, DTMF was "60 ms on, 40 ms off", way back when.  With
modern technology, shorter durations could work.  Most phones of all types
don't make a standardized tone burst but produce tones only while the button
is pressed.  Fast "punching" will produce short tones.

On the other hand, a redialed number will be very well formatted.

Reliability of TT data transfer for audio applications (over the phone voice
mail, credit card, IVR, etc) would be better if the phones would run button
pushes through the redial buffer/formatter.  But they don't.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Saturday, September 19, 2009 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]DTMF end '1' has duration 57 but want minimum
80, emulating on IAX2/a16-q1-9657

On Saturday 19 September 2009 01:07:54 Rajkumar S wrote:
> I have an occasional problem where DTMF is not recognized, ie if
> clients type a digit while in menu the system does not register it.
>
> In my C server I saw a log line like this today:
>
> DTMF end '1' has duration 57 but want minimum 80, emulating on
> IAX2/a16-q1-9657
>
> Is the above message an indication of this problem? How can I fix it?

It isn't evidence of this problem, but it might be indicative of it.  What
this message says is that the DTMF lasted for 57ms, but Asterisk normally
doesn't detect DTMF that lasts for under 80ms, so it is increasing the
duration of the DTMF to compensate (because as a digital signal, DTMF is
reliable, but when sent as audio, it might not be).  What it probably
indicates is that the DTMF sent to your system is _incredibly_ short, and if
a
DTMF detector is employed, it's possible that the DTMF audio is simply too
short to be reliably detected.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] How to stop IVR once system receives DTMF?

[Cary Fitch]

I think the IVR audio must be playing in "Background" mode, not "Play" Mode.

 

Try that.  Background means play the sound and move on to the next
instruction. Play means to play the sound and after it is over, move to the
next instruction.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B.
Reddy Bynagari
Sent: Monday, August 31, 2009 8:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to stop IVR once system receives DTMF?

 

Hi,

 

We are trying to implement a complex business logic in Asterisk. Executing
"Wait_For_Digit" command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?

 

Thanks 

BB

 

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Re: [asterisk-users] Inquiry:How to hide Caller Id

[Cary Fitch]

A Google of that model showed a discontinued Telstra corded phone.

 

But in any case SNOM and Grandstream phones Do  show the number before you
pick up the handset.

 

I would suggest you use a Grandstream 286 voip adapter and a standard corded
or wireless phone so that the caller doesn't have a display to see.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: Monday, August 31, 2009 1:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id

 

Sorry for mis-typing in phone type . Please be informed that the current
phone type our subscribers are using is "TP6000" ones .

On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales  wrote:


I couldn't find any information on this brand of phone on the internet
at all.

PaulH



hadi motamedi wrote:
> Sorry for lack of enough information . I mean my subscriber when goes
> off hook he will see his own number displayed on his phone . I need to
> disable this feature on my Asterisk .The phone type is ANABELL phone .
> Please do me favor and let me know how can I disable this feature on
> my Asterisk ?
> Looking forward your reply
> Regards
> H.Motamedi
>
>
>
> On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell  <mailto:li...@venturevoip.com>> wrote:
>
> On 31/08/09 5:24 PM, hadi motamedi wrote:
> > Dear All
> > Can you please do me favor and let me know how I can hide the subs
> > number being displayed on his phone when he goes off hook ? I
> mean when
> > the subs goes off hook he sees his assigned number on his phone
> and I
> > need to disable this feature . I don't know from which configuration
> > file this feature is coming so please let me know how can I
> disable it .
>
> You're not really giving enough information.
>
> Who sees the number?
>
> Where do they see it?
>
> What type of phone?
>
> What is a subs?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
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Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com

[Cary Fitch]

I just tried it on 3 different numbers. Dialed as 10 digits  NPANXX

 

I was told "I am sorry but you can only dial within North America..etc.

 

C Fitch

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Sunday, August 30, 2009 9:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help me testing this webphone
atwww.VisionVoIP.com

 

Thank you everyone who tested the webphone, but I haven't got input from
anybody. Most of the calls made were unfortunately unsuccessful, but I would
like to know what error you got? Did the webphone stayed in "Loading..."
state and never completed its loading, or your local firewall blocked, it,
or something else happened?

-- 
Zeeshan A Zakaria

On Sun, Aug 30, 2009 at 1:52 AM, Zeeshan Zakaria  wrote:

Greetings everyone,

I've been trying to make this java based webphone work for everybody
visiting my website, but seems like for many users it doesn't work. In order
to get a better idea what is the success rate of this webphone, I would
appreciate help from anybody who could make a few calls from it within North
America and if it doesn't work, send me what error you get, or if it works,
tell me it sounds right, no echoing etc. I am keeping calling free for now
for testing purposes.

The webphone is located at http://www.visionvoip.com

Thanks,

-- 
Zeeshan A Zakaria




-- 
Zeeshan A Zakaria

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Re: [asterisk-users] Inquiry:How to hide Caller Id

[Cary Fitch]

That doesn't happen on all phones.  Either find a way to block that
"feature" on the phone, or change phones for that location.

 

I assume you don't want the user to know that phone's local number.

 

Cary Fitch

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: Monday, August 31, 2009 1:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id

 

Thank you for your reply . Yes , he is seeing his own number on his phone
upon going off hook and before dialing any number . Can you please do me
favor and confirm if it is not a feature of Asterisk that I can disable it ?

Regards

H.Motamedi



 

On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell  wrote:

On 31/08/09 5:49 PM, hadi motamedi wrote:
> Sorry for lack of enough information . I mean my subscriber when goes
> off hook he will see his own number displayed on his phone . I need to
> disable this feature on my Asterisk .The phone type is ANABELL phone .
> Please do me favor and let me know how can I disable this feature on my
> Asterisk ?

Hi,

If he is seeing his own number on his display before he has dialed any
numbers then it is probably a feature of the phone - in which case you
need to disable it there.

If you're talking about an incoming call then it's different.

--

Cheers,

Matt Riddell
Director
___

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http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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Re: [asterisk-users] caller id problem

[Cary Fitch]

Yes, the issue(s) is/are:

1. The VOIP provider may be masking the callerID for their own cost
allocation reasons.  That is some of the issue.

2. Your Asterisk box may forward some of the regular phone line calls with
their caller ID.

3. Somehow, the number you want to use may leak through sometimes. :-)

What you need to do is put in a simple, absolute "CallerID(num) =
3216540987" type of statement before sending the call out. Make it apply to
every call no matter what.

That isn't the syntax but you get the idea. Of course you won't have true
caller ID then, but do you want "cheap" or "real"?

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan
Sent: Friday, August 07, 2009 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] caller id problem

I'm having a weird problem with CallerIDs and I can't tell if it is a 
problem with Asterisk, the telco, or the VOIP provider I'm using.

Basically, I am using Asterisk as a proxy for my cell phone. People call 
in and the call gets forwarded to my personal number. The feature on my 
phone allows for unlimited phone calls from one number, any time, for 
$7/month, so I'm saving a bundle (I use it for outgoing too). However, 
whenever somebody calls in and the call is forwarded to my regular telco 
cell number, the number is coming up different e.g. instead of 478-9987 
(made up number) it is coming in as 383-6894. Since it is now a 
"different number" I am getting charged for incoming calls and my neat 
trick is no longer working.

I'd just like to know if anybody has an inkling as to where the problem 
might be. I've tried to use Asterisk to set the CallerID and nothing has 
changed. I have called both the telco and VOIP provider's tech support 
and they both seem to blame the other.

To make things even more strange, over the course of dozens and dozens 
of calls, I have twice received a call from the correct number! That is 
the 478-9987 number, not the 383-6894. But I have no idea what the 
conditions where to make that happen.

Additionally, it seems that most everybody else who gets a call from the 
Asterisk box receives the correct number, suggesting that the problem is 
with the telco. But I can't be certain, and besides their tech support 
is no help at all. I'm running out of options and I may need to switch 
providers.

I know this is only loosely related to Asterisk, but any help would be 
greatly appreciated.

Thanks in advance.

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Re: [asterisk-users] Setting up Outgoing Trunk

[Cary Fitch]

Are these "trunks" or "PRI/ISDN circuits", or "phone lines"?

 

If either of the first two, the callerID sent with the call should be their
ID, which should be the appropriate number of digits your area telco
expects.  Depending on your agreement with them, they may be supplying the
number, rather than accept what you send.

 

If your connection is "phone lines" they are supplying the Line Number, and
you have no control over that except by strategic use of the lines, etc.

 

Or if there is further info or questions, explain the exact details.

 

Cary Fitch

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kumarshantanu
Sent: Thursday, August 06, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Setting up Outgoing Trunk

 

Hello Everybody,

I have a genuine problem in Asterisk setup.
I have three inbound trunks in my asterisk box, everything is
working fine but the only problem is when any user make an out-
going call through his/her extension it goes with same number labeled 
on this.

Can we set each of these lines to have fixed outgoing numbers
like if extn: 201 make an outgoing call the recipient should get different
no and if extn: 202 make an outgoing call the recipient should 
get different one.

Please can someone help me in this.

Thanks
Shantanu


 
<http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/sign
atureline@middle?> 

 

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