Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P

[Deepak Naidu]

So Steven, did the echo problem stopped once the Hardware echo cancellor card 
was installed out of the box, or you needed to do some configuration changes 
like Rx & Tx etc.
   
  Thanks for sharing your experience.
   
  --
  Deepak

»Steven Ringwald« <[EMAIL PROTECTED]> wrote:
  Deepak Naidu wrote:
> Hi,
> I am currently using TE110P Digium card on a PRI card. Basically 
> the echo is so much that one can disticntly identify that. I have tried 
> all the combination if tuning configuration seen in forums etc. I am 
> using MG2 cancellor algorithm & also tuned the RX & TX gains, still 
> there is an echo.
> 
> So I am thing to purchase an hardware based echo cancellor like Digium 
> Wildcard TE212P.
> 
> So in this regards I would like to get some view whether its worth to 
> buy a hardwrae based echo cancellor. Will this resolve the issue, or 
> will be just waste of money.
> 
> I am using Asterisk 1.2.18 & latest version of zaptel drivers.
> 
> Hope if someone had the same issue, I what has done to resolve it would 
> be much appreciable.
> 
In my experience, it is well worth the money. After installing several 
for customers, we never bought the non-HWEC cards again...

Steve
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RE: [asterisk-users] Confirmation key to answer -- for a queue

[Yuan LIU]

From: Yaakov Menken <[EMAIL PROTECTED]>
Date: Sun, 13 May 2007 00:59:54 -0400

Hi,

Pretty sure I'm missing something simple, but I've seen references to this 
feature but not found documentation for it:


I have a queue set up so that many people are contacted (ringall) when a 
call comes in. I would like the answering party to confirm that he is a 
human being rather than cellphone voicemaill by pressing a digit. This is 
somewhat similar to the 2nd macro example found at

http://www.voip-info.org/wiki-Asterisk+cmd+Dial


Thought it would be chanspec 'c'.  
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels


Yuan Liu


Is there a queues.conf option that I'm missing here?

Thanks for any advice,

Yaakov Menken



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[asterisk-users] Confirmation key to answer -- for a queue

[Yaakov Menken]

Hi,

Pretty sure I'm missing something simple, but I've seen references to 
this feature but not found documentation for it:


I have a queue set up so that many people are contacted (ringall) when a 
call comes in. I would like the answering party to confirm that he is a 
human being rather than cellphone voicemaill by pressing a digit. This 
is somewhat similar to the 2nd macro example found at

http://www.voip-info.org/wiki-Asterisk+cmd+Dial

Is there a queues.conf option that I'm missing here?

Thanks for any advice,

Yaakov Menken

--
Yaakov Menken
Capalon Communications, Inc.
Ask us about Voice over IP for Business!

http://www.capalon.com
888-CAPALON (227-2566)
410-358-9800 x120
410-510-1053 fax
443-413-1042 cell
[EMAIL PROTECTED]
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Re: [asterisk-users] Dry Copper Pair

[Stephen Bosch]

C F wrote:
> Stephen i disagree. growing up in new work city i can say its quite
> easy to get away with it in the city. where i live now in new jersey
> (population of around 6) i wouldnt be able to pull that off.

The world is a big place, and I suppose there's room for all kinds. In
these parts, the vigilance is pretty high. The pillars are padlocked
now; they didn't use to be, and the COs are locked down like Fort Knox.

Anyway, I know enough more than one person who has landed in the clink
for treating the telco like a personal lab.

-Stephen-
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Re: [asterisk-users] Dry Copper Pair

[Stephen Bosch]

[EMAIL PROTECTED] wrote:
> In todays socio/political climate, telco infrastructure is seen as
> foundational, and an essential service that is vital in times of
> emergency. Any unauthorised modification can present an unacceptable risk
> exposure to the telco, the emergency services, and to the public in
> general. This said, the telcos may not be providing the best security (and
> in some cases their security is non-existent) however this does not mean
> that an unauthorised person is entitle to make changes, or even enter the
> site.

In these parts, security has only gotten tighter with time. Pillars are
padlocked, and the vigilance is way up. Increased population and free
access to information on the Internet has made everything more
vulnerable, and Telus, anyway, is not ignorant of this.

Stuff I did as a teenager (don't tell my mother) I couldn't hope to get
away with now, and at my age, "I was just curious" doesn't cut it as an
excuse any longer.

-Stephen-

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Re: Now way OT: [asterisk-users] Need some help with a very simple Queestion..

[Stephen Bosch]

[EMAIL PROTECTED] wrote:
>> Stephen Bosch wrote:
>>
 Is Marmite also available in Ontario, or only Out West?
>>>
>>> As far as I know, Marmite is available all across this land, from sea to
>>> sea to sea.
>>>
>>> Three cheers for Marmite.
>>>
>> IMO most Americans have never even *heard* of Marmite, much less tasted
>> it.
>>
>> And it's quite a hoot to watch someone ingest it for the first time.
>> Always causes a surprised look.
>>
>> Someone should write a book about it--or maybe someone already has :-)
>>
>> b.
>>
> Yep, chocolate/hazelnut it aint.
> 
> FWIW, Vegemite (the better, Auzzie variety of the same brew) is now
> illegal in the US. Something to do with its containing folate and t

Is that ban still in place? My understanding is that it was temporary
(insane) and lifted...

-Stephen-
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Re: Now way OT: [asterisk-users] Need some help with Marmite

[Stephen Bosch]

Brian Capouch wrote:
> Stephen Bosch wrote:
> 
>>>
>>> Is Marmite also available in Ontario, or only Out West?
>>
>>
>> As far as I know, Marmite is available all across this land, from sea to
>> sea to sea.
>>
>> Three cheers for Marmite.
>>
> 
> IMO most Americans have never even *heard* of Marmite, much less tasted it.

Well, Marmite is available in Canada and people even eat it here.

> And it's quite a hoot to watch someone ingest it for the first time.
> Always causes a surprised look.
> 
> Someone should write a book about it--or maybe someone already has :-)

Har :)

-Stephen-
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[asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection

[Zeeshan Zakaria]

I've solved this problem. It was very easy (only if I knew how to do it
before). I changed the UDP ports, i.e.

1. In sip.conf, bindport=5070
2. In my IP Phone server settings, www.myserver.com:5070

Now it seems to be working good and I hope there'll be no more problem with
it.
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Re: [asterisk-users] Dry Copper Pair

[C F]

Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.

On 5/12/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Jon Pounder wrote:
>> On 5/11/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
>>> On Fri, 11 May 2007, C F said something to this effect:
>>>
 Not according to Verizon (in my area anyhow), We tried it and it
>>> didn't
 work. The verizon technician insisted it wasn't real PTP copper and
 therefore anything but analog voice might/should not work.
>>>What is "PTP copper"?  Unless it's an issue of gauge.  But as far as
>>> I
>>> know, it's not.  All the standard copper used for POTS can be used for a
>>> T1 from a physical point of view, other aspects of conditioning/load
>>> coils/etc/etc not withstanding.
>> You are right, but that was not what I meant, in order for one to be
>> able to provision their own T1 over a pair of copper, the line has to
>> allow all traffic over all frequencies pass thru it. Which these lines
>> do not, since they are simply not just one long copper pair simply
>> cross connected.
>
> that's what "dry copper" is supposed to be, just a cross connect between 2
> pairs out of the CO. ie not even battery, line test equipment, or anything
> else hanging off it at the CO. any restriction should be purely a function
> of the inductance/capacitance of the wire and the connections and nothing
> else - anything else and you didn't get "dry copper" in the first place.
>
>
> just out of curiousity - anyone ever hijack pairs and get away with it ?
> (do your own cross connects on the street and utilize some crossconnect
> all within one branch of F1 cable out of the CO ?)
>
> I've been tempted in the past, and know that at least around here I would
> probably get away with it for quite some time before anyone actually cared
> enough to investigate.

Jon:

Is Thorold rural?

You wouldn't get away with this for ten minutes in an urban CO. I don't
fancy spending the night in jail.

-Stephen-
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RE: [asterisk-users] Call to Skype network

[Dave Bour]

On x86 asterisk systems, there's 3 options out there, of which the
Chanskype one I've found to be the best.  It's $20 US for a single
channel personal license or $99 / per channel on a business license.  On
the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to
configure it.  I've made a couple notes too if you want, I can send
offlist (unless it's generally wanted here onlist as I don't like taking
credit for others work).
D   

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Verscheure
Sent: Saturday, May 12, 2007 9:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call to Skype network

Hi everyone,

Is it possible to call from your Asterisk server to the Skype network?
i.e., let's say I would like to call from an extension from my Asterisk
PBX machine to a Skype account, is this possible?

I did a little bit of searching and they were talking about that's only
possible with windows machines, is this true?


greetz, Tim
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[asterisk-users] Call to Skype network

[Tim Verscheure]

Hi everyone,

Is it possible to call from your Asterisk server to the Skype network?
i.e., let's say I would like to call from an extension from my
Asterisk PBX machine to a Skype account, is this possible?

I did a little bit of searching and they were talking about that's
only possible with windows machines, is this true?


greetz, Tim
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[asterisk-users] zonedata.c

[Jadrien Wauthier]

Hi,

Could anyone tell me how to read the values in the "zonedata.c" file?  I am 
looking at the "zt_tone_ringtone" field mainly.

Thank you.

Jad Wauthier

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Re: [asterisk-users] List of telemarketers??

[Dave Bour]

Suspect it is however I get one spoof a day on average. Most I recognize as 
invalid dialplan numbers though
There's a thought, a incoming dialplan validity checker... If a number can't 
pass a basic nxx-nxx-, or is from an unassigned npa, auto zap it

That said, incoming skype are often 000-012-3456

D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]

For those who just want it to work...
Giving you complete IT peace of mind. 

(Sent via Blackberry - hence message may be shorter than my usual verbose 
responses)
PIN 4cc364db (as of March 24, 2007)  

- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Sent: Sat May 12 20:54:08 2007
Subject: RE: [asterisk-users] List of telemarketers??

> 3. a list of bogus entries..so when you look at it, you know it's a
> fake phone number...one that recently came in that got me thinking
> this was 407 111 .

I don't know much about the legal position over the other side of the pond, but 
I'm pretty sure that in the UK caller ID spoofing is illegal. There's nothing 
to stop you withholding your CLI of course, but to deliberately fake someone 
else's CLI (whether it exists or otherwise) pushes you over the line.

Is the same not the case in the US?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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RE: [asterisk-users] List of telemarketers??

[Chris Bagnall]

> 3. a list of bogus entries..so when you look at it, you know it's a
> fake phone number...one that recently came in that got me thinking
> this was 407 111 .

I don't know much about the legal position over the other side of the pond, but 
I'm pretty sure that in the UK caller ID spoofing is illegal. There's nothing 
to stop you withholding your CLI of course, but to deliberately fake someone 
else's CLI (whether it exists or otherwise) pushes you over the line.

Is the same not the case in the US?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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RE: [asterisk-users] List of telemarketers??

[Dave Bour]

So a few questions...
1. Can we dump the entire database with names, category and the
collective hit count/sync anything we already have.
2. Can we upload our hits to the collective count (and immediately 0
ours, reloading our collective count number) on a periodic basis.
3. Can we upload new items we flag
4. Can we query real time something we get, ie something like the caller
id Nerd Vittles Call Trifeca lookup to see if we have a hit since we
last updated.

And now to get really smart...a peer list such that we don't overload
your server but rather randomly pull our data from a peer.  The
challenge would be to keep the hit count accurate...possible solutions
would be rather than track hit count would be to actually record
someone's "peer id such as a mac address" and the number/date/time it
was hit... It would also then show the accuracy/relevance of a number if
it's last update was 6 months ago...etc.  

The final stage would be local implementation for each user...to put
that list to meaningful use..do we simply use it as a caller id and
category...do we build a category block individually...and does a number
qualify as a multiple category 

Thanks for the work so far.
D.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Johnson
Sent: Saturday, May 12, 2007 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] List of telemarketers??

Actually have downloaded a database of over 3000 US numbers in it so
far, and have a script to return an XML result with data in it. Coding
of the page to enter data is also on the way and should be up and
working soon. We are still working on adding the names to the numbers
that we do have, so the data returns at this time is just that the
number exists, the last time it was accessed, and the count of how many
times it has been accessed.

Taking requests if there is more data that would like to be added to the
results.

Dave Bour wrote:
> As this is a thread off a topic I spun off another thread...some 
> thoughts...or a pooled database of numbers not listed..doesn't even 
> need to be telemarketers.
> 1. track the hit count.and last used..collecitvely..see if a number is

> still being used...
> 2. name against the company...
> 3. a list of bogus entries..so when you look at it, you know it's a 
> fake phone number...one that recently came in that got me thinking 
> this was 407 111 . Right away, I knew it was a 
> spammer/telemarketer since it's not a valid number...and I was on the 
> offence answering the phone...
> In fact, a possible update to the blacklist function in asterisk is a 
> database tracking of it's usage. is it a one of that hit me 
> once..never called back...
> and another issue is the charity type...what's blacklist to you may 
> not be to me, ie..clothing pickup donation charitys...they really 
> annoy me calling several times a week, but one, I don't block as 
> they're good, their caller id is accurate and they actually leave a 
> message with a number offering you to call back if you have something 
> to pick up (granted it's a recording that does that but at least it's 
> not a hangup). Some may wish to block all those... Political donation 
> type numbers are another one that irk some but not others.
> By putting names against the number and "classification"...you could 
> at least then choose to blacklist once the first call comes in, based 
> on accurate caller id info.
>
> --
> --
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Steve 
> Totaro
> *Sent:* Thursday, May 10, 2007 7:26 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* RE: [asterisk-users] List of telemarketers??
>
> Things that will make your idea difficult.
>
> Although telemarketers are supposed to send their CallerID, many if 
> not most, I see "Unkown". I believe this is illegal but I see it many 
> times a day.
>
> I worked at a company with over one thousand toll free numbers, simply

> changing outbound CallerID to another number defeats your scripts. I 
> could also change the actual DNIS of these calls, so a telemarketer 
> could easily defeat your mechanism and still be legal.
>
> I could also send any arbitrary ten digit CallerID and DNIS which 
> looks better to the callee than "Unknown" but cannot be associated to 
> a particular telemarketer.
>
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com/
> KB3OPB
>
> --
> --
>
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Ritesh

> Agrawal
> *Sent:* Wednesday, May 09, 2007 4:38 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] List of telemarketers??
>
> Thanks everyone for the responses, encouragement and offers to help.
> I will get started on this shortly and

Re: [asterisk-users] List of telemarketers??

[Ken Johnson]

Actually have downloaded a database of over 3000 US numbers in it so 
far, and have a script to return an XML result with data in it. Coding 
of the page to enter data is also on the way and should be up and 
working soon. We are still working on adding the names to the numbers 
that we do have, so the data returns at this time is just that the 
number exists, the last time it was accessed, and the count of how many 
times it has been accessed.


Taking requests if there is more data that would like to be added to the 
results.


Dave Bour wrote:
As this is a thread off a topic I spun off another thread...some 
thoughts...or a pooled database of numbers not listed..doesn't even 
need to be telemarketers.
1. track the hit count.and last used..collecitvely..see if a number is 
still being used...

2. name against the company...
3. a list of bogus entries..so when you look at it, you know it's a 
fake phone number...one that recently came in that got me thinking 
this was 407 111 . Right away, I knew it was a 
spammer/telemarketer since it's not a valid number...and I was on the 
offence answering the phone...
In fact, a possible update to the blacklist function in asterisk is a 
database tracking of it's usage. is it a one of that hit me 
once..never called back...
and another issue is the charity type...what's blacklist to you may 
not be to me, ie..clothing pickup donation charitys...they really 
annoy me calling several times a week, but one, I don't block as 
they're good, their caller id is accurate and they actually leave a 
message with a number offering you to call back if you have something 
to pick up (granted it's a recording that does that but at least it's 
not a hangup). Some may wish to block all those... Political donation 
type numbers are another one that irk some but not others.
By putting names against the number and "classification"...you could 
at least then choose to blacklist once the first call comes in, based 
on accurate caller id info.



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Steve 
Totaro

*Sent:* Thursday, May 10, 2007 7:26 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] List of telemarketers??

Things that will make your idea difficult.

Although telemarketers are supposed to send their CallerID, many if 
not most, I see “Unkown”. I believe this is illegal but I see it many 
times a day.


I worked at a company with over one thousand toll free numbers, simply 
changing outbound CallerID to another number defeats your scripts. I 
could also change the actual DNIS of these calls, so a telemarketer 
could easily defeat your mechanism and still be legal.


I could also send any arbitrary ten digit CallerID and DNIS which 
looks better to the callee than “Unknown” but cannot be associated to 
a particular telemarketer.


Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com/
KB3OPB



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ritesh 
Agrawal

*Sent:* Wednesday, May 09, 2007 4:38 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] List of telemarketers??

Thanks everyone for the responses, encouragement and offers to help.
I will get started on this shortly and circle back with you guys.
If someone has a starter list, it would help jump start the 
efforts/motivation :-)


Ritesh


On 5/9/07, *Noah Miller* <[EMAIL PROTECTED] 
> wrote:


Hi Ritesh -

> Does anyone know if there is a known list of telemarketers?
> Something like http://whocalled.us/ with an easier access?
>
> We could all benefit if there was such a thing :-)
> If there is enough interest, I could put up a database that everyone can
> benefit from.
>
> I just need some suggestions on:
> (1) Adding new numbers based on community responses (some rule to sanity
> check)
> (2) Method that everyone would prefer to access the dbase.

Wow, that's a generous offer. I like the idea of a blacklist for
telemarketers. It's bound to be more effective than an RBL for
spammers! One thing to note: this may end up being a non-US database.
Here in the US, I've experienced great success with the
www.donotcall.gov  service. If you're in the 
US and haven't signed up

for this service, I'd highly recommend it. Of course, there may be
non-telemarketer calls that it would be nice to be able to block.


- Noah
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Re: [asterisk-users] Module wctdm24xxp not found - TDM808P on debian

[Tzafrir Cohen]

On Fri, May 11, 2007 at 11:04:12AM -0300, Juliano Fernandes Schroeder wrote:
> I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp
> i get this error
> FATAL: Module wctdm24xxp not found.
> FATAL: Error running install command for wctdm24xxp
> 
> I think i have successfully compiled the zaptel drivers, 

Easy to test: run the following in the zaptel source directory: 

  ls *.ko 

> and the card
> appears when i do a lspci

The output of lspci tells you that you have a card, but is unrelated to 
the drivers you may or may not have.

> I've searched for a solution with no success. Another problem is that when i
> do a ztcfg -vv i get
> 
> Notice: Configuration file is /etc/zaptel.conf
> line 0: Unable to open master device '/dev/zap/ctl'

I suspect that you didn't even get zaptel built.

  modinfo zaptel

Which Debian is it? Which kernel version do you use?

  uname -r

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P

[»Steven Ringwald«]

Deepak Naidu wrote:

Hi,
   I am currently using TE110P Digium card on a PRI card.  Basically 
the echo is so much that one can disticntly identify that.  I have tried 
all the combination if tuning configuration seen in forums etc.  I am 
using MG2 cancellor algorithm & also tuned the RX & TX gains, still 
there is an echo.
 
So I am thing to purchase an hardware based echo cancellor like Digium 
Wildcard TE212P.
 
So in this regards I would like to get some view whether its worth to 
buy a hardwrae based echo cancellor.  Will this resolve the issue, or 
will  be just waste of money.
 
I am using Asterisk 1.2.18  & latest version of zaptel drivers.
 
Hope if someone had the same issue, I what has done to resolve it would 
be much appreciable.
 
In my experience, it is well worth the money. After installing several 
for customers, we never bought the non-HWEC cards again...


Steve
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Re: [asterisk-users] Asterisk High-Capacity Stability

[Alex Balashov]

On Sat, 12 May 2007, Atlanticnynex said something to this effect:

Thanks Alex, some great ideas. I think, however, I'm leaning towards 
Asterisk at this point- since I have quite a bit of experience there, and 
very little with SER. At this point, I'm wondering from a dimensioning 
standpoint, what kind of capacity my machine will have (Dual Core Xeon 
2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read 
the voip-info page on dimensioning and it seems theres some mixed 
feelings about Asterisk in high-capacity environments. I guess I'm 
looking for input as to whether Asterisk could handle roughly one DS3's 
worth of calls (672 calls) just doing the LCR (I've seen some pre-built 
LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd 
write my own AGI that would use a cache).


  It's really hard to say.  My personal impression from discussions I've 
witnessed on this subject is that you might be able to get away with 1/3rd 
of a DS3, and under no circumstances more.  But that doesn't mean you can't 
make this work with a few Asterisk servers and a proxy that routes requests 
to them in a round-robin and/or other load balancing fashion.  And of 
course, I could be completely wrong.


  I know that one of the main frustrations with Asterisk on the TDM side -- 
and yes, I know you're talking about SIP-to-SIP -- is that there is no way 
it can possibly scale to handle a DS3 of PRIs, and that approach at least

grants some possibility of hardware offloading.  This has led people to
have to use TNTs and other things that have DS3 adaptors as media gateways,
although finding adequate call logic / call control is not within the 
economic reach of most.


-- Alex

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[asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P

[Deepak Naidu]

Hi,
 I am currently using TE110P Digium card on a PRI card.  Basically the 
echo is so much that one can disticntly identify that.  I have tried all the 
combination if tuning configuration seen in forums etc.  I am using MG2 
cancellor algorithm & also tuned the RX & TX gains, still there is an echo.
   
  So I am thing to purchase an hardware based echo cancellor like Digium 
Wildcard TE212P.
   
  So in this regards I would like to get some view whether its worth to buy a 
hardwrae based echo cancellor.  Will this resolve the issue, or will  be just 
waste of money.
   
  I am using Asterisk 1.2.18  & latest version of zaptel drivers.
   
  Hope if someone had the same issue, I what has done to resolve it would be 
much appreciable.
   
  --
  Deepak
   

   
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Re: [asterisk-users] Fwd: SER as a Session Border Controller

[Atlanticnynex]

I've reposted to a new thread "Asterisk High Capacity Stability" as I don't
think the subject of this thread is appropriate anymore.

-kn0x

On 5/12/07, Alex Balashov <[EMAIL PROTECTED]> wrote:



On Sat, 12 May 2007, Atlanticnynex said something to this effect:

> 'SIP Redirect Proxy', which I'm understanding to just redirect the SIP
> requests to the appropriate destination based on the routing logic- and
> OpenSER is no longer involved in the call process (meaning that
> accounting can no longer be handled); but I hear their are many
different
> ways for OpenSER to behave, can this include my described configuration?

   I gather that OpenSER can be combined with rtpproxy to stay in the
media
path as well, if desired.  And really, OpenSER can behave however you
want;
it allows you full read/write modification of pretty much any SIP header.

   In terms of what module to use for this, I had a very hard time with
this
too since, believe it or not, for such a ubiquitous open-source package,
it did not offer a straightforward way of making SQL database dips without
relying on the particular schema of some module or another.

   Ultimately, I found that the 'avp' module has a little function called
avp_db_query() which can extract query results and stick them in
individual
AVP values.  This function may have just been put there for kicks, since
what AVP "really" implements is some transparent way of storing key/value
pairs.

   So, what I really end up doing for most of my intelligent routing is
something like:

---

   avp_db_query("SELECT ip_addr, port FROM customer_proxies WHERE
customer_id = $avp(S:customer_id) AND active = true",
"$avp(S:proxy_ip);$avp(S:proxy_port)");

 if(!is_avp_set("$avp(S:proxy_ip)") ||
!is_avp_set("$avp(S:proxy_port)")) {
 xlog("L_INFO", "target-das - [$ci] - Active proxy not
found.\n")
;
 sl_send_reply("404", "Not Found");
 exit;
 }

 xlog("L_INFO", "target-das - [$ci] - Resolved proxy
$avp(S:proxy_ip):$avp(S:proxy_port)\n");

 avp_pushto("$ru/domain", "$avp(S:proxy_ip):$avp(S:proxy_port)");

 xlog("L_INFO", "target-das - [$ci] - Attempting handoff to
$ru\n");

 avp_delete("$avp(S:customer_id)");
 avp_delete("$avp(S:proxy_ip)");
 avp_delete("$avp(S:proxy_port)");

---

   Seems like the simplest approach to me.

> My SIP users can't know who the Upstream Providers are, so all traffic
must
> be 'relayed' through my server, including media.

   In that case, use Asterisk as an SBC and keep it in the media path
perhaps, or tack on an RTP proxy to OpenSER, although unfortunately I can
tell you absolutely nothing about how to go about this.

> I've confirmed that my SIP users can authenticate to OpenSER via the
> RADIUS module, but 1) how do I keep track of the call detail records
> {minutes used, user info, dest. info} and report it back to RADIUS?

   A decent RADIUS module will have the capability log the accounting
information into a RADIUS accounting backend as well, since accounting is
a
key component of RADIUS.  If not, you can use something akin to what I
described above to keep the CDRs yourself.

> how do can OpenSER authenticate to my Upstream Providers?

   I am not sure if it can.  There may have to be a B2BUA somewhere
further
upstream within your "backend platform," as is not infrequently done.

   In other implementations I've seen, all connections to providers are
done
on a hard-coded port/IP-mapped "peer" basis so that proxies can hand calls
off straight to them.  But another way is to put another SBC-type unit at
the egress to the provider(s) to take care of that.  Just beware that this
comes with the full implications of a B2BUA--distinct logical call legs.

Best of luck,

-- Alex

--
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[asterisk-users] Asterisk High-Capacity Stability

[Atlanticnynex]

Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since I have
quite a bit of experience there, and very little with SER. At this point,
I'm wondering from a dimensioning standpoint, what kind of capacity my
machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan
to do any transcoding. I read the voip-info page on dimensioning and it
seems theres some mixed feelings about Asterisk in high-capacity
environments. I guess I'm looking for input as to whether Asterisk could
handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've
seen some pre-built LCR apps, looks like they all do on-the-fly MySQL
queries- I think I'd write my own AGI that would use a cache).


With my hardware, could Asterisk run stable for this amount of traffic?
What stability issues does Asterisk have at this scale?

On 5/12/07, Alex Balashov <[EMAIL PROTECTED]> wrote:



On Sat, 12 May 2007, Atlanticnynex said something to this effect:

> 'SIP Redirect Proxy', which I'm understanding to just redirect the SIP
> requests to the appropriate destination based on the routing logic- and
> OpenSER is no longer involved in the call process (meaning that
> accounting can no longer be handled); but I hear their are many
different
> ways for OpenSER to behave, can this include my described configuration?

   I gather that OpenSER can be combined with rtpproxy to stay in the
media
path as well, if desired.  And really, OpenSER can behave however you
want;
it allows you full read/write modification of pretty much any SIP header.

   In terms of what module to use for this, I had a very hard time with
this
too since, believe it or not, for such a ubiquitous open-source package,
it did not offer a straightforward way of making SQL database dips without
relying on the particular schema of some module or another.

   Ultimately, I found that the 'avp' module has a little function called
avp_db_query() which can extract query results and stick them in
individual
AVP values.  This function may have just been put there for kicks, since
what AVP "really" implements is some transparent way of storing key/value
pairs.

   So, what I really end up doing for most of my intelligent routing is
something like:

---

   avp_db_query("SELECT ip_addr, port FROM customer_proxies WHERE
customer_id = $avp(S:customer_id) AND active = true",
"$avp(S:proxy_ip);$avp(S:proxy_port)");

 if(!is_avp_set("$avp(S:proxy_ip)") ||
!is_avp_set("$avp(S:proxy_port)")) {
 xlog("L_INFO", "target-das - [$ci] - Active proxy not
found.\n")
;
 sl_send_reply("404", "Not Found");
 exit;
 }

 xlog("L_INFO", "target-das - [$ci] - Resolved proxy
$avp(S:proxy_ip):$avp(S:proxy_port)\n");

 avp_pushto("$ru/domain", "$avp(S:proxy_ip):$avp(S:proxy_port)");

 xlog("L_INFO", "target-das - [$ci] - Attempting handoff to
$ru\n");

 avp_delete("$avp(S:customer_id)");
 avp_delete("$avp(S:proxy_ip)");
 avp_delete("$avp(S:proxy_port)");

---

   Seems like the simplest approach to me.

> My SIP users can't know who the Upstream Providers are, so all traffic
must
> be 'relayed' through my server, including media.

   In that case, use Asterisk as an SBC and keep it in the media path
perhaps, or tack on an RTP proxy to OpenSER, although unfortunately I can
tell you absolutely nothing about how to go about this.

> I've confirmed that my SIP users can authenticate to OpenSER via the
> RADIUS module, but 1) how do I keep track of the call detail records
> {minutes used, user info, dest. info} and report it back to RADIUS?

   A decent RADIUS module will have the capability log the accounting
information into a RADIUS accounting backend as well, since accounting is
a
key component of RADIUS.  If not, you can use something akin to what I
described above to keep the CDRs yourself.

> how do can OpenSER authenticate to my Upstream Providers?

   I am not sure if it can.  There may have to be a B2BUA somewhere
further
upstream within your "backend platform," as is not infrequently done.

   In other implementations I've seen, all connections to providers are
done
on a hard-coded port/IP-mapped "peer" basis so that proxies can hand calls
off straight to them.  But another way is to put another SBC-type unit at
the egress to the provider(s) to take care of that.  Just beware that this
comes with the full implications of a B2BUA--distinct logical call legs.

Best of luck,

-- Alex

--
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Re: Now way OT: [asterisk-users] Need some help with a very simple Queestion..

[thg]

> Stephen Bosch wrote:
>
>>>
>>>Is Marmite also available in Ontario, or only Out West?
>>
>>
>> As far as I know, Marmite is available all across this land, from sea to
>> sea to sea.
>>
>> Three cheers for Marmite.
>>
>
> IMO most Americans have never even *heard* of Marmite, much less tasted
> it.
>
> And it's quite a hoot to watch someone ingest it for the first time.
> Always causes a surprised look.
>
> Someone should write a book about it--or maybe someone already has :-)
>
> b.
>
Yep, chocolate/hazelnut it aint.

FWIW, Vegemite (the better, Auzzie variety of the same brew) is now
illegal in the US. Something to do with its containing folate and that
being a restricted chemical that needs FDA approval, etcetera...

So travelling Aussies, can't even bring their own little jar with them...


Hilarious thing is that Vegemite whilst invented by an Aussie, is owned
and  produced by the Kraft Company (US Company). So its ok for a US
company to make an illegal product, Make money from an illegal product,
and export an illegal product, but some one that buys that product for
their own use, cannot even bring a 4oz jar of it with them on holiday.

talk about a separation between church^Wlogic and state^Wpractice. go figure


T

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Re: [asterisk-users] Fwd: SER as a Session Border Controller

[Alex Balashov]

On Sat, 12 May 2007, Atlanticnynex said something to this effect:

'SIP Redirect Proxy', which I'm understanding to just redirect the SIP 
requests to the appropriate destination based on the routing logic- and 
OpenSER is no longer involved in the call process (meaning that 
accounting can no longer be handled); but I hear their are many different 
ways for OpenSER to behave, can this include my described configuration?


  I gather that OpenSER can be combined with rtpproxy to stay in the media 
path as well, if desired.  And really, OpenSER can behave however you want;

it allows you full read/write modification of pretty much any SIP header.

  In terms of what module to use for this, I had a very hard time with this 
too since, believe it or not, for such a ubiquitous open-source package,

it did not offer a straightforward way of making SQL database dips without
relying on the particular schema of some module or another.

  Ultimately, I found that the 'avp' module has a little function called
avp_db_query() which can extract query results and stick them in individual
AVP values.  This function may have just been put there for kicks, since
what AVP "really" implements is some transparent way of storing key/value
pairs.

  So, what I really end up doing for most of my intelligent routing is 
something like:


---

  avp_db_query("SELECT ip_addr, port FROM customer_proxies WHERE 
customer_id = $avp(S:customer_id) AND active = true",

"$avp(S:proxy_ip);$avp(S:proxy_port)");

if(!is_avp_set("$avp(S:proxy_ip)") ||
   !is_avp_set("$avp(S:proxy_port)")) {
xlog("L_INFO", "target-das - [$ci] - Active proxy not 
found.\n")

;
sl_send_reply("404", "Not Found");
exit;
}

xlog("L_INFO", "target-das - [$ci] - Resolved proxy 
$avp(S:proxy_ip):$avp(S:proxy_port)\n");


avp_pushto("$ru/domain", "$avp(S:proxy_ip):$avp(S:proxy_port)");

xlog("L_INFO", "target-das - [$ci] - Attempting handoff to $ru\n");

avp_delete("$avp(S:customer_id)");
avp_delete("$avp(S:proxy_ip)");
avp_delete("$avp(S:proxy_port)");

---

  Seems like the simplest approach to me.


My SIP users can't know who the Upstream Providers are, so all traffic must
be 'relayed' through my server, including media.


  In that case, use Asterisk as an SBC and keep it in the media path 
perhaps, or tack on an RTP proxy to OpenSER, although unfortunately I can 
tell you absolutely nothing about how to go about this.


I've confirmed that my SIP users can authenticate to OpenSER via the 
RADIUS module, but 1) how do I keep track of the call detail records 
{minutes used, user info, dest. info} and report it back to RADIUS?


  A decent RADIUS module will have the capability log the accounting 
information into a RADIUS accounting backend as well, since accounting is a 
key component of RADIUS.  If not, you can use something akin to what I 
described above to keep the CDRs yourself.



how do can OpenSER authenticate to my Upstream Providers?


  I am not sure if it can.  There may have to be a B2BUA somewhere further 
upstream within your "backend platform," as is not infrequently done.


  In other implementations I've seen, all connections to providers are done 
on a hard-coded port/IP-mapped "peer" basis so that proxies can hand calls

off straight to them.  But another way is to put another SBC-type unit at
the egress to the provider(s) to take care of that.  Just beware that this
comes with the full implications of a B2BUA--distinct logical call legs.

Best of luck,

-- Alex

--
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Re: [asterisk-users] Fwd: SER as a Session Border Controller

[Atlanticnynex]

Thanks Alex, for the quick and detailed response. I really appreciate it.
Well, that's pretty much what I've been getting from my reading.
I'm curious as to what areas (specifically, modules/features) of OpenSER I
should look into to accomplish this.
Essentially I don't want any endpoint to endpoint traffic or visibility
(both endpoints only see the 'backend platform' as their endpoint) for SIP
and RTP.
I understand the most common OpenSER configuration is that it acts only as a
'SIP Redirect Proxy', which I'm understanding to just redirect the SIP
requests to the appropriate destination based on the routing logic- and
OpenSER is no longer involved in the call process (meaning that accounting
can no longer be handled); but I hear their are many different ways for
OpenSER to behave, can this include my described configuration? Also, just
to clarify,  I do not plan to transcode anything- or any other molestation
of the media stream.
I'm sure there must be some way to do this all, as I hear a great number of
VoIP carriers have employed (Open)SER and other open source software to run
their networks.

So in short I this is what I'm looking to do:


(SIP Users)--[
   ]  /- (Upstream SIP Provider)
(SIP Users)--[  My Switch/ Backend
Platform  ] 0 --(Upstream SIP Provider)
(SIP Users)--[   (LCR trunk selection, CDR, outbound
auth.) ]  \--(Upstream SIP Provider)

My SIP users can't know who the Upstream Providers are, so all traffic must
be 'relayed' through my server, including media.
I've confirmed that my SIP users can authenticate to OpenSER via the RADIUS
module, but 1) how do I keep track of the call detail records {minutes used,
user info, dest. info} and report it back to RADIUS? and 2) how do can
OpenSER authenticate to my Upstream Providers? I've been pointed to the
'UAC' module, but how can I integrate that with the LCR  module (each
'gateway' / provider has different authentication info. and the LCR module
mentions nothing about outbound authentication)? It is very important that
the SIP Users are not relayed the outbound authentication info for security
purposes.

Thanks,

kn0x


On 5/12/07, Alex Balashov <[EMAIL PROTECTED]> wrote:



Greetings,

   It is my impression that Asterisk cannot safely handle more than about
100-200 calls in parallel, but it may be possible to increase the yield by
removing any transcoding and offloading some of the channel functionality
to hardware DSP boards.  I do not know much about this, and specifically
do
not know how much help they would be if there is no transcoding to be
performed per se.

   Some of the stuff you're wanting to do - authenticating via RADIUS,
perform an LCR lookup, select trunks - is best done by a proxy that sits
behind Asterisk.  OpenSER can certainly interface with this kind of call
logic, and what's more, it can act as a registrar.  It certainly does have
a bit of an abstruse learning curve, but I can say it's not impossible to
figure out by any stretch of the imagination, having been through that
multiple times for inbound call processing / distribution tasks.

   I am not sure that what you are attempting to describe really fits the
definition of a session border controller, however.  An SBC really only
holds the SIP URI reachability information (contacts) for end-users and
not much else.  In most other respects it behaves rather like a proxy;
authentication is done by handoff to a backend registrar/proxy, any kind
of call routing is also typically farmed out to backend proxies.  The
SBC more or less does exactly what its name suggests;  it provides a
"border," a logical horizon of call control.  Otherwise, it's a pretty
dumb device, whereas what you appear to be alluding to sounds more like
a rather intelligent processing endpoint.

   Sometimes the SBC stays in the media path, and sometimes it
doesn't.  It
depends on the implementation.  And some SBCs can provide primitive native
registrars, but this isn't typically thought scalable or desirable from a
systems integration standpoint.

   So, in the grand scheme of things, the situation usually resembles:

  +--+
  | Backend  |
 Client <--/- Internet -/> SBC <> | Platform |
  |  |
  +--+

   A "backend platform" can be an array of specialised proxies and
registrars, which may or may not be one and the same, that ultimately
direct the call to other endpoints within or outside of the service
provider's network.  Or it can be a softswitch that has a built-in
call agent / registrar / proxy / media gateway rolled in, or any number
of other curious configuration possibilities.

-- Alex


--
Alex Balashov   <[EMAIL PROTECTED]>
__

Re: Now way OT: [asterisk-users] Need some help with a very simple Queestion..

[Brian Capouch]

Stephen Bosch wrote:



Is Marmite also available in Ontario, or only Out West?



As far as I know, Marmite is available all across this land, from sea to
sea to sea.

Three cheers for Marmite.



IMO most Americans have never even *heard* of Marmite, much less tasted it.

And it's quite a hoot to watch someone ingest it for the first time. 
Always causes a surprised look.


Someone should write a book about it--or maybe someone already has :-)

b.

--
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believed to be clean.

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[asterisk-users] Recent zaptel versions break CLIP?

[Remco Barendse]

Hi!

Is it just me or do the last 2 or 3 versions of the zaptel-1.2 branch seem 
to break cli? Often not the full number is displayed, or only 2 or 3 
digits?


I am in The Netherlands, and have had this in my zapata.conf (which used 
to work flawlessly) :

signalling=fxs_ks
immediate=yes
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no


I installed some new boxes with newer zaptel cards recently, but same 
problem.


Thanks for any hints / tips!
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Re: [asterisk-users] Dry Copper Pair

[Paul]

Eric "ManxPower" Wieling wrote:

> Jon Pounder wrote:
>
>> that's what "dry copper" is supposed to be, just a cross connect
>> between 2
>> pairs out of the CO. ie not even battery, line test equipment, or
>> anything
>> else hanging off it at the CO. any restriction should be purely a
>> function
>> of the inductance/capacitance of the wire and the connections and
>> nothing
>> else - anything else and you didn't get "dry copper" in the first place.
>>
>>
>> just out of curiousity - anyone ever hijack pairs and get away with it ?
>> (do your own cross connects on the street and utilize some crossconnect
>> all within one branch of F1 cable out of the CO ?)
>>
>> I've been tempted in the past, and know that at least around here I
>> would
>> probably get away with it for quite some time before anyone actually
>> cared
>> enough to investigate.
>
>
> At least in Bellsouth/Louisiana they do not guarantee that the circuit
> will pass DC voltage.   Since it is an alarm circuit I believe they
> only  guarantee that it will pass short/open.  If the circuit goes
> between COs then I there is no reason for them to pass DC voltage.  If
> it is within the same CO then there is no reason I can think of that
> it would not pass DC voltage, except of course to prevent people from
> using xDSL tech on the line.

A little history for the youngsters:

I remember when I was 7(that would be 1959), there were several false
alarms sent from the fire alarm box on the nearby street corner. I
watched the process of resetting the alarm box. It was much like
rewinding a clock. They swept up the pieces of broken glass on the
sidewalk and then installed a new glass pane in the alarm box.

The basic operation was that you pulled a handle down which broke
through the glass pane and triggered an unwinding of the clock spring
mechanism. That mechanism was basically a telegraph pulse sender. I was
standing close enough when they tested the box to hear the soft growl of
a spring driven motor and the clicking of the telegraph switch.

I don't know what the sending rate was for those devices. Whatever it
was, compare it to the bit rates we can now get over the same dry pair.
I think back then there were very few people who would believe that 40
years later(1999) over a megabit per second would be common.

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Re: [asterisk-users] Dry Copper Pair

[thg]

> Quoting Stephen Bosch <[EMAIL PROTECTED]>:
>
>> Jon Pounder wrote:
 On 5/11/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
> On Fri, 11 May 2007, C F said something to this effect:
>
>> Not according to Verizon (in my area anyhow), We tried it and it
> didn't
>> work. The verizon technician insisted it wasn't real PTP copper and
>> therefore anything but analog voice might/should not work.
>What is "PTP copper"?  Unless it's an issue of gauge.  But as far
> as
> I
> know, it's not.  All the standard copper used for POTS can be used
> for a
> T1 from a physical point of view, other aspects of conditioning/load
> coils/etc/etc not withstanding.
 You are right, but that was not what I meant, in order for one to be
 able to provision their own T1 over a pair of copper, the line has to
 allow all traffic over all frequencies pass thru it. Which these lines
 do not, since they are simply not just one long copper pair simply
 cross connected.
>>>
>>> that's what "dry copper" is supposed to be, just a cross connect
>>> between 2
>>> pairs out of the CO. ie not even battery, line test equipment, or
>>> anything
>>> else hanging off it at the CO. any restriction should be purely a
>>> function
>>> of the inductance/capacitance of the wire and the connections and
>>> nothing
>>> else - anything else and you didn't get "dry copper" in the first
>>> place.
>>>
>>>
>>> just out of curiousity - anyone ever hijack pairs and get away with it
>>> ?
>>> (do your own cross connects on the street and utilize some crossconnect
>>> all within one branch of F1 cable out of the CO ?)
>>>
>>> I've been tempted in the past, and know that at least around here I
>>> would
>>> probably get away with it for quite some time before anyone actually
>>> cared
>>> enough to investigate.
>>
>> Jon:
>>
>> Is Thorold rural?
>>
>> You wouldn't get away with this for ten minutes in an urban CO. I don't
>> fancy spending the night in jail.
>>
>
> it not a big city, but its not the middle of nowhere either.
>
> The local CO here is actually the last hop on the fibre trunks from
> Canada to the US for the Toronto area, fence around it has the gate
> open 24x7, so its not just the local loops that they don't really care
> enough to protect.
>
> I have seen techs find things they didn't agree with or think should
> be somewhere and just shrug their shoulders - if its not a reported
> problem they don't want to be the one that touches something and
> breaks it if its really supposed to be there. A lot of the techs are
> subcontracted by the job, they don't get paid to make improvements,
> they get paid to do installations and go on repair calls, so if its
> not on their work order they just don't care.
>
> Here is an example, I order lines in batches of 1 or 2 at a time to my
> location, every time an installer comes they put another 2pr aerial
> cable from my pole to the pole across the street. I have plenty of
> underground capacity I put in myself out to my pole. Everytime I say,
> hey why not just consolidate things in a properly sized aerial cable
> or just bury it ? No, can't do that, all I can do is install yet
> another cable (there are about 10 up there now, and it looks like
> hell). I have even got the response, "well if you want to clean it up
> you could just do it and no one would care". If that's not a direct
> invitation to work on the telco outside plant yourself - what is ?
>
>
Uh-huh, so... say, leaving my car unlocked while it is sitting in my
driveway is "directly invitation" to someone to take it?

The exchange and all their feeder sites, rims, pillars, etc... are the
property of the telco, at the very least working on them without explicit
authority is a breech of the tresspass laws and it goes up from there.

You will probably find that they (the telco) will take the point of view
that (and you can confirm this with any lawyer you like) their failure to
prosecute every instance of tresspass does not imply permission to enter.

In todays socio/political climate, telco infrastructure is seen as
foundational, and an essential service that is vital in times of
emergency. Any unauthorised modification can present an unacceptable risk
exposure to the telco, the emergency services, and to the public in
general. This said, the telcos may not be providing the best security (and
in some cases their security is non-existent) however this does not mean
that an unauthorised person is entitle to make changes, or even enter the
site.

Any argument to the contrary is somewhat shortsighted, dont you think?

Or is your failure to lock your front door, a "direct invitation" to the
next person that walks down the sidewalk to help themselves to the
contents of your fridge?

It could be argued that some of this thread constitutes a solicitation to
commit a crime, or perhaps coersion of the same. In the US, it wouldnt
surprise me if tresspass into a telco pillar would get 

Re: [asterisk-users] Dry Copper Pair

[Jon Pounder]

Quoting "Eric \ManxPower\ Wieling" <[EMAIL PROTECTED]>:


Jon Pounder wrote:


that's what "dry copper" is supposed to be, just a cross connect between 2
pairs out of the CO. ie not even battery, line test equipment, or anything
else hanging off it at the CO. any restriction should be purely a function
of the inductance/capacitance of the wire and the connections and nothing
else - anything else and you didn't get "dry copper" in the first place.


just out of curiousity - anyone ever hijack pairs and get away with it ?
(do your own cross connects on the street and utilize some crossconnect
all within one branch of F1 cable out of the CO ?)

I've been tempted in the past, and know that at least around here I would
probably get away with it for quite some time before anyone actually cared
enough to investigate.


At least in Bellsouth/Louisiana they do not guarantee that the circuit
will pass DC voltage.   Since it is an alarm circuit I believe they
only  guarantee that it will pass short/open.


how can you "pass" a short/open without passing dc ?

alarm circuits are normally an always on low speed modem that I have  
ever seen in practice - I know back in the stone age there was such a  
thing as you describe, but I don't think its been in use since at  
least the 70's or 80's. The alarm application wouldn't have to pass  
dc, just ac, BUT if it doesn't pass DC its not "dry copper" its  
probably what is being referred to as the "class A channel" which is  
just an end to end connection for audio but no dialtone. ie its  
possible to pass a modem signal on either type, but the channel might  
do FX by being aggregated to fibre, etc as a 64k channel between COs  
but the dry copper could never be done that way.






 If the circuit goes

between COs then I there is no reason for them to pass DC voltage.  If
it is within the same CO then there is no reason I can think of that it
would not pass DC voltage, except of course to prevent people from
using xDSL tech on the line.
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Re: [asterisk-users] muscionhold error message

[Jonathan Addleman]

Tzafrir Cohen wrote:
> It actually is (maintained, and a recent version of it is in 
> stable/testing.

Hmm.. I think several years ago it wasn't... I guess I'm just living in
the past. Sorry about that!
-- 
Jon-o Addleman - http://www.redowl.ca
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Re: [asterisk-users] Dry Copper Pair

[Jon Pounder]

Quoting Stephen Bosch <[EMAIL PROTECTED]>:


Jon Pounder wrote:

On 5/11/07, Alex Balashov <[EMAIL PROTECTED]> wrote:

On Fri, 11 May 2007, C F said something to this effect:


Not according to Verizon (in my area anyhow), We tried it and it

didn't

work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but analog voice might/should not work.

   What is "PTP copper"?  Unless it's an issue of gauge.  But as far as
I
know, it's not.  All the standard copper used for POTS can be used for a
T1 from a physical point of view, other aspects of conditioning/load
coils/etc/etc not withstanding.

You are right, but that was not what I meant, in order for one to be
able to provision their own T1 over a pair of copper, the line has to
allow all traffic over all frequencies pass thru it. Which these lines
do not, since they are simply not just one long copper pair simply
cross connected.


that's what "dry copper" is supposed to be, just a cross connect between 2
pairs out of the CO. ie not even battery, line test equipment, or anything
else hanging off it at the CO. any restriction should be purely a function
of the inductance/capacitance of the wire and the connections and nothing
else - anything else and you didn't get "dry copper" in the first place.


just out of curiousity - anyone ever hijack pairs and get away with it ?
(do your own cross connects on the street and utilize some crossconnect
all within one branch of F1 cable out of the CO ?)

I've been tempted in the past, and know that at least around here I would
probably get away with it for quite some time before anyone actually cared
enough to investigate.


Jon:

Is Thorold rural?

You wouldn't get away with this for ten minutes in an urban CO. I don't
fancy spending the night in jail.



it not a big city, but its not the middle of nowhere either.

The local CO here is actually the last hop on the fibre trunks from  
Canada to the US for the Toronto area, fence around it has the gate  
open 24x7, so its not just the local loops that they don't really care  
enough to protect.


I have seen techs find things they didn't agree with or think should  
be somewhere and just shrug their shoulders - if its not a reported  
problem they don't want to be the one that touches something and  
breaks it if its really supposed to be there. A lot of the techs are  
subcontracted by the job, they don't get paid to make improvements,  
they get paid to do installations and go on repair calls, so if its  
not on their work order they just don't care.


Here is an example, I order lines in batches of 1 or 2 at a time to my  
location, every time an installer comes they put another 2pr aerial  
cable from my pole to the pole across the street. I have plenty of  
underground capacity I put in myself out to my pole. Everytime I say,  
hey why not just consolidate things in a properly sized aerial cable  
or just bury it ? No, can't do that, all I can do is install yet  
another cable (there are about 10 up there now, and it looks like  
hell). I have even got the response, "well if you want to clean it up  
you could just do it and no one would care". If that's not a direct  
invitation to work on the telco outside plant yourself - what is ?







-Stephen-
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Jon Pounder

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_/_/_/  _/  _/ _/_/_/  _/  _/_/
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_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
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[asterisk-users] Preparing music on hold

[Chris Bagnall]

Greetings list,

I've been having a go at preparing some music on hold from CDs clients have 
supplied, but quality seems really rather poor over compressed channels (tried 
g729, GSM and Speex). I've been doing the following:

sox -v 0.15  -t raw -r 8000 -s -w -c 1  resample -ql

As I understand it, I'm reducing volume to 15% (which sounds about right 
volume-wise) and downsampling to 8khz, 16-bit mono.

I know MoH over compressed links isn't ideal conditions and is never going to 
be great, but I should be able to at least make it bearable.

What's the prevailing opinion on using high and low pass filters? One would 
assume a phone handset is expected to provide frequency response in human 
speech zones, and not really much outside that (certainly not the 20hz-20khz 
one might expect of a CD).

Suggestions gratefully appreciated.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons




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Re: [asterisk-users] Dry Copper Pair

[Eric \"ManxPower\" Wieling]

Jon Pounder wrote:


that's what "dry copper" is supposed to be, just a cross connect between 2
pairs out of the CO. ie not even battery, line test equipment, or anything
else hanging off it at the CO. any restriction should be purely a function
of the inductance/capacitance of the wire and the connections and nothing
else - anything else and you didn't get "dry copper" in the first place.


just out of curiousity - anyone ever hijack pairs and get away with it ?
(do your own cross connects on the street and utilize some crossconnect
all within one branch of F1 cable out of the CO ?)

I've been tempted in the past, and know that at least around here I would
probably get away with it for quite some time before anyone actually cared
enough to investigate.


At least in Bellsouth/Louisiana they do not guarantee that the circuit 
will pass DC voltage.   Since it is an alarm circuit I believe they only 
 guarantee that it will pass short/open.  If the circuit goes between 
COs then I there is no reason for them to pass DC voltage.  If it is 
within the same CO then there is no reason I can think of that it would 
not pass DC voltage, except of course to prevent people from using xDSL 
tech on the line.

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[asterisk-users] DTMF detection problem on wctdm24xxp

[Paradise Dove]

hi all,
i have problem with dtmf detection on wctdm24xxp with full fxo and vpm module.
after pushing dtmf tones on my phone for several times the card just
detects one or two digits randomly.so now i can't use any voice menu
on my box with this card.

i have tried the following scenarios:

- the card with / without vpm module has the same dtmf detection problem.
- relaxdtmf=yes/no didn't solve the problem
- toneduration=300 / 350 / 400 didn't help also.
- vpmdtmfsupport=1 / 0 didn't solve again.

what else could be the possible cause for this problem?

please help!
- paradise dove
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Re: [asterisk-users] Dry Copper Pair

[thg]

> Jon Pounder wrote:
>>> On 5/11/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
 On Fri, 11 May 2007, C F said something to this effect:

> Not according to Verizon (in my area anyhow), We tried it and it
 didn't
> work. The verizon technician insisted it wasn't real PTP copper and
> therefore anything but analog voice might/should not work.
What is "PTP copper"?  Unless it's an issue of gauge.  But as far
 as
 I
 know, it's not.  All the standard copper used for POTS can be used for
 a
 T1 from a physical point of view, other aspects of conditioning/load
 coils/etc/etc not withstanding.
>>> You are right, but that was not what I meant, in order for one to be
>>> able to provision their own T1 over a pair of copper, the line has to
>>> allow all traffic over all frequencies pass thru it. Which these lines
>>> do not, since they are simply not just one long copper pair simply
>>> cross connected.
>>
>> that's what "dry copper" is supposed to be, just a cross connect between
>> 2
>> pairs out of the CO. ie not even battery, line test equipment, or
>> anything
>> else hanging off it at the CO. any restriction should be purely a
>> function
>> of the inductance/capacitance of the wire and the connections and
>> nothing
>> else - anything else and you didn't get "dry copper" in the first place.
>>
>>
>> just out of curiousity - anyone ever hijack pairs and get away with it ?
>> (do your own cross connects on the street and utilize some crossconnect
>> all within one branch of F1 cable out of the CO ?)
>>
>> I've been tempted in the past, and know that at least around here I
>> would
>> probably get away with it for quite some time before anyone actually
>> cared
>> enough to investigate.
>
> Jon:
>
> Is Thorold rural?
>
> You wouldn't get away with this for ten minutes in an urban CO. I don't
> fancy spending the night in jail.
>
> -Stephen-

In some places a night in jail is getting off real easy. Try 3 months in
jail and then $multi thousands as a fine, and if you were acting on behalf
of a corporation, then $millions as a fine.

It is not misdemeanor issues, it is criminal.

T

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[asterisk-users] 1.2.17 -> 1.2.18 asterisk crash

[marek cervenka]

hi,

i am updated to latest asterisk stable (because of security problems), but 
now asterisk crashes within a hour


log is clear

do you someone have this problem too?

---
Marek Cervenka
===

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Re: [asterisk-users] Rapid DTMF missing digits

[Bryan Laird]

Yea that was my first guess until I saw the packet dump prove out  
that the ATA was transmitting it.  Eh, let me go searching through  
the bug lists see if I can find something in older versions.


On May 11, 2007, at 3:04 PM, Matt wrote:

I have actually seen this behaviour on 1.2.x.   I always assumed it  
was just me dialing too fast for the ATA.


On 5/11/07, Bryan Laird < [EMAIL PROTECTED]> wrote:
Version 1.4.2 but to be honest I have no reason at all to suspect
that this is a problem with the asterisk software.

I've able to replicate this from a few different "client" net
connections and a across a few different linksys ata's.  Where when
you call into the
host and enter the extension to connect to you miss the last digit of
the extension.  Almost every time you miss the last digit of the
extension
(in a 4 digit extension).  My suspicion is simply because of the
network we are currently using to host the asterisk box, as a packet
dump on the
lan segment clearly showed that the ATA transmitted all digits
(rfc2833) but the asterisk host only recieved 3 of the 4.  The second
you dial
slower everything works fine; also the lines for "voice" are clear
with no noticeable impairments.  I'm more curious if anyone else has
ever run
into a similar problem and what the resolution was if they found one
(IE a sturdier net connection for the asterisk host),  or Tweaking
the timers
on the ata's to slow down how fast and how long they transmit
digits.  I've done a few different tests and if I use a 'softphone'
dialing directly into
the server things work perfectly.  I can dial as fast as I want,
however when I come in through the pstn trunks through the upstream
provider I find this problem.

has anyone else ever seen this?  Or seen a case where mis-matched
dtmf modes across multiple providers causes this problem?

minor detail on what I referred to as the 'pstn trunks' I have no
analog or digital circuts all handoffs are sip.


-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird
Saving Lost Packets since 1994
Have you seen this packet? 101010010
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-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird, Sr. Manager CM Operations
Phone: 703-944-9909
   -+-
Cablemodems are the gateway to the Internet.
The Internet is a gateway to some things that are  better left un-seen.


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[asterisk-users] Remote extensions not working on provider's wireless Internet connection

[Zeeshan Zakaria]

Hi,

My SIP phones which are working fine at all other remote locations, when
placed in a location connected to the Internet through Rogers Wireless
Internet, they don't work. They do get registered on low port numbers, like
, 1123, 1231 etc. but can't dial out or receive calls. Asterisk CLI show
no activity for them. Its the same for ATAs and IP phones. I've tried all
the settings, played aroung with NAT, but to no avail. Can't make them
register on higher port numbers. Same devices work fine if connected
anywhere else.

What could be the reason and how to solve it?

--
Zeeshan A Zakaria
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[asterisk-users] ser problem

[Pezhman Lali]

Dear
I am using ser + asterisk, for setting up land line
calling.
only probelm, each  unregistered soft phone can places
the call only with callerid,
this is critical problem, because any number(soft
phone) , has a limit time to use this system,
best
Mani




   
Be
 a better Globetrotter. Get better travel answers from someone who knows. 
Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469
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Re: [asterisk-users] muscionhold error message

[Tzafrir Cohen]

On Fri, May 11, 2007 at 11:08:50PM -0400, Jonathan Addleman wrote:
> pedro noticioso wrote:
> > hi there guys!
> > 
> > how can I eliminate this message?
> > 
> > [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
> > monmp3thread: Unable to spawn mp3player
> > [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
> > spawn_mp3: Found no files in
> > '/var/lib/asterisk/mohmp3'
> 
> I'm no expert, but haven't seen other replies here, so I'll throw out my
> suggestions.
> 
> I don't have that package (still using testing's 1.2 package) but I
> expect the /etc/asterisk/musiconhold.conf uses mp123 to process the
> music on hold files. That's not Free software though (and isn't really
> maintained anymore, I think...), 

It actually is (maintained, and a recent version of it is in 
stable/testing.

However editing musiconhold.conf shouldn't be a problem. I wonder if 
we should start distributing a separate musiconhold package, as the 
current one is non-free: if you happen to make the mistake of Playback of 
those FPM sound files, you're violating their license.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Fwd: SER as a Session Border Controller

[Alex Balashov]

Greetings,

  It is my impression that Asterisk cannot safely handle more than about 
100-200 calls in parallel, but it may be possible to increase the yield by

removing any transcoding and offloading some of the channel functionality
to hardware DSP boards.  I do not know much about this, and specifically do
not know how much help they would be if there is no transcoding to be
performed per se.

  Some of the stuff you're wanting to do - authenticating via RADIUS, 
perform an LCR lookup, select trunks - is best done by a proxy that sits

behind Asterisk.  OpenSER can certainly interface with this kind of call
logic, and what's more, it can act as a registrar.  It certainly does have
a bit of an abstruse learning curve, but I can say it's not impossible to
figure out by any stretch of the imagination, having been through that
multiple times for inbound call processing / distribution tasks.

  I am not sure that what you are attempting to describe really fits the 
definition of a session border controller, however.  An SBC really only

holds the SIP URI reachability information (contacts) for end-users and
not much else.  In most other respects it behaves rather like a proxy;
authentication is done by handoff to a backend registrar/proxy, any kind
of call routing is also typically farmed out to backend proxies.  The
SBC more or less does exactly what its name suggests;  it provides a 
"border," a logical horizon of call control.  Otherwise, it's a pretty

dumb device, whereas what you appear to be alluding to sounds more like
a rather intelligent processing endpoint.

  Sometimes the SBC stays in the media path, and sometimes it doesn't.  It 
depends on the implementation.  And some SBCs can provide primitive native

registrars, but this isn't typically thought scalable or desirable from a
systems integration standpoint.

  So, in the grand scheme of things, the situation usually resembles:

 +--+
 | Backend  |
Client <--/- Internet -/> SBC <> | Platform |
 |  |
 +--+

  A "backend platform" can be an array of specialised proxies and 
registrars, which may or may not be one and the same, that ultimately

direct the call to other endpoints within or outside of the service
provider's network.  Or it can be a softswitch that has a built-in
call agent / registrar / proxy / media gateway rolled in, or any number
of other curious configuration possibilities.

-- Alex


--
Alex Balashov   <[EMAIL PROTECTED]>
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