[asterisk-users] ISDN, overlap and open dialing plans (Olivier)

Tue Mar 13 12:53:50 CDT 2012

Dear Oliver

Well I never knew PBX could according to numbers pattern length  use enbloc and overlap, from my experiences dialling mode setting should be one or other, but it should be set on whole system one mode, by the way if number length pattern is variable component to use overlap mode.
Just in case when it use enbloc PBX send  whole number,  TDM phones it  press # or other key setting as send digits.
By the other way check dialplan rules to resolve receiving number length, a good practice is use and Asterisk extension to simulated call from PBX system.
Best regards

Mc GRATH Ricardo
E-Mail mcgrathr at mail2web.com
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Today's Topics:

   1. ISDN, overlap and open dialing plans (Olivier)
   2. DAHDI_SPANCONFIG failed on span 1: No such device or address
      (6) (RaMaier at gmx.de)
   3. Re: Capacity of single instance of Asterisk
      (Amit Patkar | Avhan Technologies Pvt Ltd)
   4. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
      address (6) (Eric Wieling)
   5. Re: Capacity of single instance of Asterisk
      (Amit Patkar | Avhan Technologies Pvt Ltd)
   6. Re: how to show used "wrong password" (Kevin P. Fleming)
   7. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
      address (6) (Shaun Ruffell)
   8. Re: Capacity of single instance of Asterisk (Kevin P. Fleming)
   9. Re: Capacity of single instance of Asterisk (Bryant Zimmerman)
  10. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
      address (6) (Tzafrir Cohen)
  11. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
      address (6) (RaMaier at gmx.de)
  12. Re: how to show used "wrong password" (Randall)
  13. Re: how to show used "wrong password" (Randall)

----------------------------------------------------------------------

Message: 1
Date: Tue, 13 Mar 2012 14:37:06 +0100
From: Olivier <oza_4h07 at yahoo.fr>
Subject: [asterisk-users] ISDN, overlap and open dialing plans
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users at lists.digium.com>
Message-ID:
        <CAPeT9jjSovFSbyJyeHbSB1si0KNn_OcyUnmHm+=O-_TCrwuHtg at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Hi,

I've got the following setup:

PSTN/ISDN <---- E1-----> Asterisk  <---- E1-----> Alcatel 4400 PBX
<----> TDM phones

When a TDM phone is dialing out to a national number, it seems that
the PBX is using enbloc dialing.
When a TDM phone is dialing out to an international number (variable
length numbers), it seems that the PBX is using overlap dialing as
Asterisk is currently receiving truncated numbers.

What is the best way to deal with such situations ?
1. configure PSTN in enbloc dialing and tweak dialplan to mimic
overlap dialing ?
2. or configure both PTSN and PBX spans in overlap mode ?
Suggestions ?

Regards

------------------------------

Message: 2
Date: Tue, 13 Mar 2012 15:30:45 +0100
From: RaMaier at gmx.de
Subject: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such
        device or address (6)
To: asterisk-users at lists.digium.com
Message-ID: <20120313143045.18460 at gmx.net>
Content-Type: text/plain; charset="utf-8"

Hi all,

I have problems starting dahdi.
dahdi_cfg -vvv allwasy comes back with:

DAHDI Tools Version - 2.2.1.1

DAHDI Version: 2.3.0.1
Echo Canceller(s):
Configuration
======================

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02)
Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03)

3 channels to configure.

DAHDI_SPANCONFIG failed on span 1: No such device or address (6)

I searched the internet but could not yet find a solution.
I already tried to exchange the zaphfc drivers as suggested, but they did not compile.

I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card.

Any suggestions / hints / tutorials / links welcome.

Do I need some special drivers in the kernel ?
Modprobe ?
Anything else special I need ?

Thanks
Rainer

Details:

I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules.

I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel.
download Asterisk -> ./configure -> make -> make install -> make samples
download dahdi.tar.bz2 -> make -> make install

dahdi_hardware:
driver should be 'zaphfc' but is actually 'hfcpci'
pci:0000:00:0d.0     zaphfc+      1397:2bd0 HFC-S ISDN BRI card

--
NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone!
Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

------------------------------

Message: 3
Date: Tue, 13 Mar 2012 20:09:26 +0530
From: "Amit Patkar | Avhan Technologies Pvt Ltd" <amit at avhan.com>
Subject: Re: [asterisk-users] Capacity of single instance of Asterisk
To: <asterisk-users at lists.digium.com>
Message-ID: <20120313143824.2F75F444FE at corpmail.avhan.com>
Content-Type: text/plain;       charset="us-ascii"

Hi Steve

Thanks for your input. Please check my comments.

> I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How
> many concurrent SIP sessions I can run from single instance of
> Asterisk on this server? I wish to use G711 codec with echo cancel.
> And all calls needs to be recorded.

What kind of capacity are you looking to achieve?

[Amit Patkar] Some where 2400 G.711 sessions with recording. So approx 1200
calls.

>From my experience, Asterisk is not really much of a RAM hog. A couple
>GB
is good for a couple hundred simultaneous calls.

With 4 'Intel(R) Xeon(TM) CPU 3.40GHz' cores, I can handle a couple hundred
simultaneous non-transcoding calls with no recording on Asterisk 1.2.

With 24 cores and 16 GB on tap, you will probably find other resource
limitations before either CPU or RAM are a limiting factor.

Personally, I'm a 'don't put all your eggs in one basket' kind of guy.

Assuming a simplistic linear relationship between my host and yours, what
will you do when it crashes with 1600 calls in progress? What will you do
when you need to install patches or upgrade or ...

I like a couple of instances of OpenSIPS in front of several Asterisk
instances, even if OpenSIPS is on the same boxes as Asterisk.

[Amit Patkar] I completely agree with you on distributing the load. At the
same time, I am looking at juicing hardware as well. Can you share the
number instead of saying couple hundreds?

> What will be impact on no of session when G729a is used?

Significant.

[Amit Patkar] Can I assume 30% reduction? Or it would be much more.

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

------------------------------

Message: 4
Date: Tue, 13 Mar 2012 10:40:08 -0400
From: Eric Wieling <EWieling at nyigc.com>
Subject: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No
        such device or address (6)
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users at lists.digium.com>
Message-ID:
        <C262B52114110B4586FAF49F074F05801083FC736F at mailserver2007.nyigc.globe>

Content-Type: text/plain; charset="us-ascii"

This means the config file says 3 ports, but no card is detected.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of RaMaier at gmx.de
Sent: Tuesday, March 13, 2012 10:31 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)

Hi all,

I have problems starting dahdi.
dahdi_cfg -vvv allwasy comes back with:

DAHDI Tools Version - 2.2.1.1

DAHDI Version: 2.3.0.1
Echo Canceller(s):
Configuration
======================

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03)

3 channels to configure.

DAHDI_SPANCONFIG failed on span 1: No such device or address (6)

I searched the internet but could not yet find a solution.
I already tried to exchange the zaphfc drivers as suggested, but they did not compile.

I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card.

Any suggestions / hints / tutorials / links welcome.

Do I need some special drivers in the kernel ?
Modprobe ?
Anything else special I need ?

Thanks
Rainer

Details:

I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules.

I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel.
download Asterisk -> ./configure -> make -> make install -> make samples download dahdi.tar.bz2 -> make -> make install

dahdi_hardware:
driver should be 'zaphfc' but is actually 'hfcpci'
pci:0000:00:0d.0     zaphfc+      1397:2bd0 HFC-S ISDN BRI card

--
NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone!
Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

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------------------------------

Message: 5
Date: Tue, 13 Mar 2012 20:13:55 +0530
From: "Amit Patkar | Avhan Technologies Pvt Ltd" <amit at avhan.com>
Subject: Re: [asterisk-users] Capacity of single instance of Asterisk
To: <asterisk-users at lists.digium.com>
Message-ID: <20120313144252.BB24C444FE at corpmail.avhan.com>
Content-Type: text/plain;       charset="us-ascii"

Hi Kevin,

Thank for your views. Where as no one is ready to share real numbers. I am
looking at benchmarks so that I can plan for resources.
Since asterisk project is active for so many years, I was expecting some
published numbers.

Thanks & Regards,
Amit Patkar

On 03/12/2012 03:38 PM, Steve Edwards wrote:
> On Mon, 12 Mar 2012, Amit Patkar wrote:

>> What will be impact on no of session when G729a is used?

Assuming that transcoding is involved; if all the system is doing is
passing through G.729A media streams, and recording them in unmixed
G.729A format, there's no additional impact (the system might actually
perform slightly better, as there is substantially less data being
shuffled around).

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

------------------------------

Message: 6
Date: Tue, 13 Mar 2012 09:53:15 -0500
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [asterisk-users] how to show used "wrong password"
To: asterisk-users at lists.digium.com
Message-ID: <4F5F5F5B.4050008 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 03/13/2012 08:11 AM, A J Stiles wrote:
> On Tuesday 13 March 2012, Randall wrote:
>> hi all,
>>
>> have asterisk set up in combination with fail2ban.
>> all works as expected only there is 1 extension that is trying to
>> register with a wrong password causing fail2ban to block the IP address,
>> normally that is ok behaviour but i have several extensions on that IP
>> address.
>> ..... snip .....
>> anyway to see which "wrong password" is being used?
>
> tcpflow.
>
> (And don't underestimate the power of simply disconnecting things until it
> works .....  last thing you disconnected was the faulty one.)

This will not help. Assuming we are talking about a SIP REGISTER here,
the password is *not* sent in the request. Asterisk issues a challenge
including a randomly generated value (called a 'nonce'), then the UA
attempting to register responds to that challenge with an MD5 digest of
a string composed of various elements, including both the nonce and the
shared secret ('password'). Asterisk computes the same digest
internally, and if they match, then the assumption is that both ends
know the shared secret.

By their very nature, digest functions are not reversible; given the MD5
digest present in an SIP request containing an Authorization header,
there is no way to figure out what shared secret was used in the
computation of that digest. Since you know the nonce and the other
portions of the calculation, you could attempt to try various 'likely'
passwords to see if any of them result in the same digest value... this
is called the brute-force method, and it could take a *very* long time
to arrive at a shared secret that would allow the endpoint to register.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

------------------------------

Message: 7
Date: Tue, 13 Mar 2012 09:47:51 -0500
From: Shaun Ruffell <sruffell at digium.com>
Subject: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No
        such device or address (6)
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users at lists.digium.com>
Message-ID: <20120313144751.GA16433 at digium.com>
Content-Type: text/plain; charset=us-ascii

On Tue, Mar 13, 2012 at 03:30:45PM +0100, RaMaier at gmx.de wrote:
>
> dahdi_hardware:
> driver should be 'zaphfc' but is actually 'hfcpci'
> pci:0000:00:0d.0     zaphfc+      1397:2bd0 HFC-S ISDN BRI card

Looks like you may need to blacklist hfcpci in
/etc/modprobe.d/blacklist.conf.

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

------------------------------

Message: 8
Date: Tue, 13 Mar 2012 10:06:34 -0500
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [asterisk-users] Capacity of single instance of Asterisk
To: asterisk-users at lists.digium.com
Message-ID: <4F5F627A.6090308 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 03/13/2012 09:43 AM, Amit Patkar | Avhan Technologies Pvt Ltd wrote:

> Thank for your views. Where as no one is ready to share real numbers. I am
> looking at benchmarks so that I can plan for resources.
> Since asterisk project is active for so many years, I was expecting some
> published numbers.

You have completely missed the point that other posters have made
already on this list. Let me try to express it another way. Let's say
that you were browsing at an engine manufacturer's website, looking at
V-8 gasoline engines, and you found one that you liked, that you felt
had a good combination of features for your project. If you then
contacted the manufacturer and asked them 'how fast can this engine make
a car travel', what do you think their response would be?

Asterisk is a toolkit; it can be configured an infinite number of ways.
Any performance measurements that are made and published apply *only* to
the specific configuration that was measured; it may or may not be
possible to extrapolate those into other configurations, or higher/lower
capacities.

There are lots of published numbers of Asterisk being used in various
ways and for different purposes; whether any of them apply to your
specific project is debatable, and relying on them for your project
would carry some level of risk. Whether you are willing to accept that
risk or not is up to you.

In your specific case, as has been mentioned already, it is extremely
unlikely that your proposed hardware would have any trouble with
Asterisk 1.8 handling 2,400 SIP call legs (1,200 bridged calls), with
the same codec being used on both sides. When you add in transcoding,
that will change the system significantly, and depending on the codecs
involved, the hardware may still be able to handle the load. I know from
experiments I did years ago with an 8-core Xeon machine (2nd generation
Xeon, so nowhere near as powerful as modern Xeon cores) that the Digium
G.729 codec (software implementation) could handle over 800 channels
with Asterisk 1.4; I think it's reasonable to expect that given the
hardware you've proposed, transcoding 1,200 channels between G.711 ulaw
and G.729A is likely to be achievable.

Recording, though, is an entirely different matter. Again, since you
haven't provided specifics, let's assume you are going to record the
call legs 'as is' (in their native formats, unmixed). If you had 2,400
G.711 ulaw call legs to record, some simple math says that you'd need be
able to push 150 megabytes per second of data onto your filesystem, on
top of all the 'normal' work that Asterisk is doing. That's rather a
lot, and will require that your filesystem and disk subsystem be
extremely fast and well tuned.

If the call legs were all G.729A, then the amount of data to write would
drop to 18.75 megabytes per second, which is achievable even with
inexpensive SATA disks.

If you want the calls recorded in 'mixed' form (most likely in 16-bit
signed linear PCM audio, since that's the easiest format to use outside
of Asterisk), you'd double the amount of data going into the filesystem
(now 300 megabytes per second) *and* you'd add in the CPU consumption of
having to decode the incoming media streams and mix them. For G.711 ulaw
this is pretty cheap and would likely not be an issue; for G.729A it's
somewhat more expensive, but still might not be a problem given the
amount of CPU capacity you have proposed.

Now do you understand why 'benchmarks' don't provide much value for
something like Asterisk?

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

------------------------------

Message: 9
Date: Tue, 13 Mar 2012 11:06:20 -0400
From: "Bryant Zimmerman" <BryantZ at zktech.com>
Subject: Re: [asterisk-users] Capacity of single instance of Asterisk
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
        <asterisk-users at lists.digium.com>
Message-ID: <21b7b9b1$2fcd9184$3cdee209$@com>
Content-Type: text/plain; charset="us-ascii"

----------------------------------------
 From: "Kevin P. Fleming" <kpfleming at digium.com>
Sent: Tuesday, March 13, 2012 11:02 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Capacity of single instance of Asterisk

On 03/13/2012 09:43 AM, Amit Patkar | Avhan Technologies Pvt Ltd wrote:

> Thank for your views. Where as no one is ready to share real numbers. I
am
> looking at benchmarks so that I can plan for resources.
> Since asterisk project is active for so many years, I was expecting some
> published numbers.