Re: [asterisk-users] Asterisk - Nortel transfer problem

2012-04-24 Thread Jonn Taylor 

You need to change it to QSIG or this will continue to be a problem.

On 04/24/2012 12:05 PM, Carlos Chavez wrote:

The E1 between the Asterisk and Nortel is using R2 for signalling.  The
PSTN comes to Asterisk first and then send calls to the Nortel.  When we
started we were just replacing an automatic operator/voicemail system
for the Nortel and all calls went there.  The customer has been
gradually shifting extensions to Asterisk and plans to phase out the
Nortel completely by next year so we will see this problem crop up more
often.

On Tue, 2012-04-24 at 11:45 -0500, Jonn Taylor wrote:

Please post your E1 configs. If you are not using QSIG you should. On
the nortel side this only works well with R6.0 and later. I have a
simular setup but with Cisco UCM but the calls come into the Nortel
first and then can be passed back and forth between them with no problem.

On 04/24/2012 10:39 AM, Carlos Chavez wrote:

I have an Asterisk server connected to a Nortel Pbx via an E1.
Everything works fine, I get calls in and out with callerid.  The
problem that has been reported to me is the following scenario:

A call comes in from the PSTN and is answered by Asterisk.  The person
dials the operator (1000) which is on the Nortel side so connection is
made through the E1.  The operator answers and then transfers the call
back to a SIP extension on the Asterisk (1303).  The result is no audio
and a dropped call.

My main theory at the moment is that when the receptionist hangs up
after the transfer the E1 drops on the Nortel side.  Anyone here with
this type of integration seen this problem?



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Re: [asterisk-users] Asterisk - Nortel transfer problem

2012-04-24 Thread Jonn Taylor 
Please post your E1 configs. If you are not using QSIG you should. On 
the nortel side this only works well with R6.0 and later. I have a 
simular setup but with Cisco UCM but the calls come into the Nortel 
first and then can be passed back and forth between them with no problem.


On 04/24/2012 10:39 AM, Carlos Chavez wrote:

I have an Asterisk server connected to a Nortel Pbx via an E1.
Everything works fine, I get calls in and out with callerid.  The
problem that has been reported to me is the following scenario:

A call comes in from the PSTN and is answered by Asterisk.  The person
dials the operator (1000) which is on the Nortel side so connection is
made through the E1.  The operator answers and then transfers the call
back to a SIP extension on the Asterisk (1303).  The result is no audio
and a dropped call.

My main theory at the moment is that when the receptionist hangs up
after the transfer the E1 drops on the Nortel side.  Anyone here with
this type of integration seen this problem?



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Re: [asterisk-users] Reliable SIP Trunk Provider

2012-03-15 Thread Jonn Taylor 
I have been using bandwidth.com since 2006 and have no problems at all. 
They do not support t38 but have free local termination in a lot of US 
cities. Tech support is good and they do support asterisk.


Jonn

On 03/15/2012 10:45 AM, Jake Wicke wrote:
I'm wondering if any other Asterisk users have a recommendation for a 
reliable SIP Trunk provider that supports Asterisk and offers decent 
support.
I've worked with Coredial, Broadvox, and Broadvoice and have had some 
bad experiences with each of these providers.
Broadvoice offers low cost service, however I have constant issues 
with Broadvoice blocking my customers due to Asterisk "registering too 
often".  Support either does not respond to e-mails, hangs up on phone 
calls, or gives me the "we don't support Asterisk and we can use your 
account no problem using the SIP phone on our desk" line.
Coredial resigned me into a two year agreement after making a change 
to my SIP trunk configuration without my knowledge, then demanded two 
years of the full monthly charge when I tried to cancel over a dispute 
regarding services that I did not order.  Check out 
coredialhorrorstory.com for the whole story.  While the service is 
decent, the customer service leaves much to be desired.
Broadvox has been the best provider that I have found so far, however 
I initially had a lot of issues with sales quoting a product which 
could not be provisioned and also not being able to deliver service on 
a timely schedule.  I also was given the run around by customer 
service recently on a simple request to add a DID number to an account.

Thanks for your input!


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Re: [asterisk-users] Avaya 4610sw IP Phone

2012-01-23 Thread Jonn Taylor 

On 01/23/2012 08:28 AM, Aamir Chougule wrote:


Hi All,

Does anyone know how to directly make an Avaya 4610sw IP phone to 
communicate directly with the Asterisk server.


Regards,

PPT

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This phone only works with Avaya IP Ofiice.

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[asterisk-users] Digium TDM 400 or Openvox A400P

2009-05-14 Thread Jonn Taylor 
What is the difference between these to cards? Any feed back good or bad 
would be great.

Jonn

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Re: [asterisk-users] DID's in a specific rate center

2009-02-25 Thread Jonn Taylor 

http://en.wikipedia.org/wiki/Caller_ID_spoofing

Danny Nicholas wrote:

Depends on the purpose.  If I'm representing a client in another state with
their permission, it's perfectly legit for me to spoof their number.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk
Sent: Wednesday, February 25, 2009 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID's in a specific rate center

Danny Nicholas wrote:
  

If you're using them outgoing only, you should consider "spoofing" the
number (IE calling using XXX-XXX- and presenting as 916-854-).


This
  

would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's.

  


You do know that that's illegal, right?

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Re: [asterisk-users] zaptel telephone cards and asterisk in another pc

2009-02-20 Thread Jonn Taylor 

Are your pc's linux or windows?

Ignacio wrote:

Jeff I will take a more depth look at those linksys devices this
weekend but I think they could be very interesting.

Tzafrir, what I like to avoid is installing an asterisk server in
every user computer. I think that is useless I want only one server to
mantain.

On Fri, Feb 20, 2009 at 7:55 PM, Tzafrir Cohen  wrote:
  

On Fri, Feb 20, 2009 at 07:11:04PM +0100, Ignacio wrote:


Thank you very much for your fast answer Eric.

I was trying to avoid to have to install as many asterisk as pcs I
have. But I think there is no way to do it. I only have seen something
like network block device, but not sure if it is going to work and
quite difficult to configure properly.

Anyway I think the fast and easier way will be installing and asterisk
in every client.
  

I guess you can use TDMoE. But I'm not really sure it will give you a
lower overhead.

Specifically, why is it that you want to avoid installing Asterisk
there? The requirements of an Asterisk system for a few analog channels
and a few uncompressed SIP/IAX channels are rather minimal.

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

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Re: [asterisk-users] life safety system and VOIP

2009-02-17 Thread Jonn Taylor 

John Novack wrote:

Jonn Taylor wrote:
  

Jon Pounder wrote:


Don E. Wisdom wrote:
  
  

On 2/17/09 2:05 PM, "Jon Pounder"  wrote:

Jeff LaCoursiere wrote:
> What do you suppose we have as liability if we are asked to
install such
> systems? Is it the responsibility of the business owner that
orders the
> system to meet all applicable codes? If (god forbid) someone was
hurt in
> such a situation and the alarm didn't get passed because of being
> delivered by VoIP for whatever reason, does the system installer
have any
> liability?
>

>well here's a question - which is more reliable ?
>- a single copper line dialed on demand when there is a problem
>- voip or other internet technology, using internet connections on
more
>than one media (say phone and cable), voip connected to multiple
servers
>in a failover configuration.

>its not uncommon for even a house to have multiple internet
connections,
>but how many buildings have phone lines that connect back to different
>CO's and fail over ?

>The best bet if you really care about what you are trying to
protect is
>make sure the message can get out as many ways as possible, whether it
>be phone, voip, network, cellmodem, etc. Forget what regulations
>require, no one says you can't go further than the minimum if you
want.

In a REAL emergency internet/cell is more likely to fail than the
phone companys pots network.
Cable/DSLAM etc only have about 4 hours of battery power. The CO
has a entire battery room which will last a whole lot longer. Not
to mention that it may stay up longer than your VoIP network. You
also have to take into account everything between you& the CO or
cable company. If just ONE thing fails you loose voip. Copper is a
lot more forgiving & has failover modes versus the phone co’s ATM
network or the cable companies “network” (or lack there of)

--Don



I don't know if thats really true any more, all the new areas around 
here have satellite CO's where fibre comes out to a box on the street 
with some batteries etc and copper runs out from there - great for dsl 
since its close, but at the mercy of whatever batteries are in there.
  
  
The dial tone for the phone line still comes from the CO. The phone 
companies loop there copper cable in and out of the remote cabinets.

Obviously you are unaware of the very many SLIC cabinets and vaults in 
use in the US.

Fewer and fewer "dial tone" comes directly from the CO.
He is correct. These are remote D to A converters that are at the mercy 
of the batteries in the remotes, some last 4 hours, if they are 
maintained. In other areas the Telco's have to scramble with portable 
generators to keep service up. In other cases even the CO's can't 
outlast the devastation of an ice storm, and have to have power brought 
in, all assuming the local Telco is able to.
  
I am very aware of how the public telephone network works as our company 
installs CO's for many different telephone companies all over the US. 
Yes some of them install all of the equipment in the remote cabinets and 
others do not. Some do fiber to home. They all have batteries that can 
fail.

maybe your alarm needs to report in since there is a fire in your phone 
equipment - what then ?

I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact 
of life.
  
  

This is true, that is why most fire panels have to have 2 phone lines.



True, but when both lines are served from the same CO, over the same 
cable, it is really a false sense of security.
In the US also, dry copper supervised pairs are scarce as hens teeth any 
more. Time was a copper pair was supervised with a DC current from end 
to end, and if something would open the circuit, that alerted the 
monitoring station there was a trouble. If there was a real alarm, they 
DC was reversed, and the monitoring station would react accordingly. 
Ancient history now. Dry pairs have disappeared over the last 20-30 
years, and many other schemes have come and gone.
  
Not true!!!  The telephone companies today are driven by money. They 
still can provide dry pairs. They just do not want to, its not in their 
best interest.
Few UL and NFPA systems allow VOIP though. Risk management still 
considers it unreliable, and of course, they are correct.
Anyone who believes otherwise, ask your business insurance provider for 
a ruling.
  


This is very true. Anyone ever read the disclaimer from vonage?

John Novack

  





  
  



> j
>
> On Tue, 17 Feb 2009, Jason Aarons (US) wrote:
&

Re: [asterisk-users] life safety system and VOIP

2009-02-17 Thread Jonn Taylor 

Jon Pounder wrote:

Don E. Wisdom wrote:
  


On 2/17/09 2:05 PM, "Jon Pounder"  wrote:

Jeff LaCoursiere wrote:
> What do you suppose we have as liability if we are asked to
install such
> systems? Is it the responsibility of the business owner that
orders the
> system to meet all applicable codes? If (god forbid) someone was
hurt in
> such a situation and the alarm didn't get passed because of being
> delivered by VoIP for whatever reason, does the system installer
have any
> liability?
>

>well here's a question - which is more reliable ?
>- a single copper line dialed on demand when there is a problem
>- voip or other internet technology, using internet connections on
more
>than one media (say phone and cable), voip connected to multiple
servers
>in a failover configuration.

>its not uncommon for even a house to have multiple internet
connections,
>but how many buildings have phone lines that connect back to different
>CO's and fail over ?

>The best bet if you really care about what you are trying to
protect is
>make sure the message can get out as many ways as possible, whether it
>be phone, voip, network, cellmodem, etc. Forget what regulations
>require, no one says you can't go further than the minimum if you
want.

In a REAL emergency internet/cell is more likely to fail than the
phone companys pots network.
Cable/DSLAM etc only have about 4 hours of battery power. The CO
has a entire battery room which will last a whole lot longer. Not
to mention that it may stay up longer than your VoIP network. You
also have to take into account everything between you& the CO or
cable company. If just ONE thing fails you loose voip. Copper is a
lot more forgiving & has failover modes versus the phone co’s ATM
network or the cable companies “network” (or lack there of)

--Don




I don't know if thats really true any more, all the new areas around 
here have satellite CO's where fibre comes out to a box on the street 
with some batteries etc and copper runs out from there - great for dsl 
since its close, but at the mercy of whatever batteries are in there.
  
The dial tone for the phone line still comes from the CO. The phone 
companies loop there copper cable in and out of the remote cabinets.
maybe your alarm needs to report in since there is a fire in your phone 
equipment - what then ?


I have seen every type of media go down or have problems no matter how 
stable - the only answer is have more than one so you always have a 
backup. Poles get hit, cables get cut, equipment breaks, its just a fact 
of life.
  

This is true, that is why most fire panels have to have 2 phone lines.








  






> j
>
> On Tue, 17 Feb 2009, Jason Aarons (US) wrote:
>
>
>>
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650
>> ;p=1
>>
>>
>> I can't see the Dept Transportation running copper to all the
motorist
>> aid boxes along the highway. I thought most of your alarm panels
have
>> moved to GSM/CDMA backup communications. I'd like to see a fire
>> marshall not give a permit for having a VoIP ATA or Vonage.
>>
>>
>>
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650
>> ;p=1
>>
>> It's permitted in Chapter 8 2002 & 2007 "Alternative Methods of
>> Communication" and these still have supervision in accordance
with Chap
>> 4 and it's sub-section.
>>
>> 8.5.2.2* Alternate Methods.
>> 8.5.4 Other Transmission Technologies.
>>
>> 8.6.2.2* Alternate Methods.
>> 8.6.4 Other Transmission Technologies.
>>
>> There is nothing specific with regards to voice over internet
protocal
>> and leaves room to add new technology proposals with requirements in
>> future editions according to A8.5.2.2. or A8.6.2.2 respectively.
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
    >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
>> LaCoursiere
>> Sent: Tuesday, February 17, 2009 3:28 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Credit Card processing machines
>>
>>
>>
>> On Tue, 17 Feb 2009, Jonn Taylor wrote:
>>
>>
>>> If you are in the US, ANY life safety system has to be
connected to a
>>> dedicated copper

Re: [asterisk-users] Credit Card processing machines

2009-02-17 Thread Jonn Taylor 

Jeff LaCoursiere wrote:

On Tue, 17 Feb 2009, Jerry Jones wrote:

  

Most alarm systems around here use bursts of dtmf - not an actual
modem to communicate with alarm central.

Yes I have seen these have many issues with voip in the path.




You mean they communicate with an IVR?  Seems like that could be made 
solid with the right DTMF options enabled on the ATA.


FWIW that makes a lot more sense than a modem connection.

j

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If you are in the US, ANY life safety system has to be connected to a 
dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the 
NFPA.



Jonn
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Re: [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call

2009-02-06 Thread Jonn Taylor 
Giorgio Incantalupo wrote:
> Hi,
>
> just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same 
> zaptel/libpri/mISDN/add-ons.
> It crashes when transferring a call.
> Anybody tried it with success?
>
> Thank you
>
> Giorgio
>
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You really should upgrade all of them. But you have to do add-ons!

-- 
Jonn Taylor

Taylor Telephone Systems, Inc
8334 Argenta Trail
Inver Grove Heights, MN 55077

http://www.taylortelephone.com/



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[asterisk-users] Problem with MOH and streaming music on 1.6.0.5

2009-02-04 Thread Jonn Taylor 
I am having a problem getting MOH to work with mpg123 on 1.6. I created 
a bug ticket
 and I am not getting any where so I am looking here for help.

Please see http://bugs.digium.com/view.php?id=14387 for details.

-- 
Jonn Taylor

Taylor Telephone Systems, Inc
8334 Argenta Trail
Inver Grove Heights, MN 55077
http://www.taylortelephone.com/



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Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???

2009-01-31 Thread Jonn Taylor 
Jonn Taylor wrote:
> Doug Lytle wrote:
>   
>> Jonn Taylor wrote:
>>   
>> 
>>> Anyone else having problems connecting to 
>>> http://downloads.digium.com/pub/ ??
>>>   
>>> 
>>>   
>> It would appear it isn't down, but it's not responding to http requests.
>>
>> Doug
>>
>>
>>   
>> 
> Yes, That is the same thing that I am getting.
>
> Jonn
>
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Ok, it just started working again.

Jonn

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Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???

2009-01-31 Thread Jonn Taylor 
Doug Lytle wrote:
> Jonn Taylor wrote:
>   
>> Anyone else having problems connecting to 
>> http://downloads.digium.com/pub/ ??
>>   
>> 
>
> It would appear it isn't down, but it's not responding to http requests.
>
> Doug
>
>
>   
Yes, That is the same thing that I am getting.

Jonn

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[asterisk-users] Is http://downloads.digium.com/pub/ down???

2009-01-31 Thread Jonn Taylor 
Anyone else having problems connecting to 
http://downloads.digium.com/pub/ ??

Jonn

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Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released

2009-01-30 Thread Jonn Taylor 

Noah Miller wrote:

The policy that we have been following is that only final releases will be
announced to the asterisk-announce list. Betas and release candidates are not.
The rationale is that asterisk-announce is supposed to be a low-volume list and
that most subscribers to it would not appreciate all the "noise" of announcing
release candidates or betas there.

I should think that the policy could be amended; however, I'm not really in a
position to make that call, nor do I know if you're a vocal minority or if most
subscribers to the -announce list would appreciate seeing such messages.



Survey?  I would appreciate such postings to the -announce list.  Even
with the rc release notices, it will still be a very low volume list.


- Noah

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I second that.

--
Jonn Taylor

Taylor Telephone Systems, Inc
http://www.taylortelephone.com/


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Re: [asterisk-users] Web page to show online extensions?

2008-04-03 Thread Jonn Taylor 
Its called Flash Operator Panel or FOP. It is install with freepbx, but 
I think you can use it as a standalone app.

Jonn

Vincent wrote:
> Hello
>
> Has someone written a web page (preferably PHP) that simply shows what
> extensions are currently online?
>
> Thank you.
>
>
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Re: [asterisk-users] CentPBX mirror?

2008-04-02 Thread Jonn Taylor 
I have a some setup scripts that use centos 4 or 5 and freepbx you are 
welcome to use them.

Jonn

http://www.taylortelephone.com/asterisk/


Chris Bagnall wrote:
>> CentPBX has bit the dust I believe.
>> 
>
> Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel 
> support for a Dell R200 (it's usually the SAS controller that causes the 
> problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather 
> too "customized" for what I'm after for this deployment.
>
> TIA.
>
> Regards,
>
> Chris
>   

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Re: [asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jonn Taylor 
Eric "ManxPower" Wieling wrote:
> Any echo you hear on pure IP calls is caused by the endpoint phone.  You 
> cannot do ANYTHING about it on Asterisk.
>
>
> Jonn Taylor wrote:
>   
>> Any ideas ?
>>
>> Jonn
>>
>>  Original Message 
>> Subject: [asterisk-users] Internal LAN echo problem
>> Date:Wed, 24 Oct 2007 08:34:32 -0500
>> From:Jonn R Taylor <[EMAIL PROTECTED]>
>> Reply-To:Asterisk Users Mailing List - Non-Commercial Discussion 
>> 
>> To:  Asterisk Users Mailing List - Non-Commercial Discussion 
>> 
>>
>>
>>
>> Hi all,
>>
>> I have an internal echo problem on my LAN only. I replaced the LAN 
>> switch with a new linksys 2024 with QOS and seemed to help but not fix 
>> the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
>> Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
>> an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
>> cheap that are known for echo problem in the handset. I have one remote 
>> user that never has a problem. I have a remote test server at home 
>> connect via IAX with no problems, also a PAP2 with no problem. External 
>> faxing from the rest of the world via our voip provider is working 
>> great. One strange thing that I noticed is that we can not fax to our 
>> iaxmodem, ATA ---> iaxmodem, but works perfect ATA ---> rx_fax. Not sure 
>> why either.
>> 
>
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That does not make sense. I can any one of these ata's or phones and 
connect them to the public ip side and they work fine.

Jonn

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[asterisk-users] [Fwd: Internal LAN echo problem]

2007-10-24 Thread Jonn Taylor 
Any ideas ?

Jonn

 Original Message 
Subject:[asterisk-users] Internal LAN echo problem
Date:   Wed, 24 Oct 2007 08:34:32 -0500
From:   Jonn R Taylor <[EMAIL PROTECTED]>
Reply-To:   Asterisk Users Mailing List - Non-Commercial Discussion 

To: Asterisk Users Mailing List - Non-Commercial Discussion 




Hi all,

I have an internal echo problem on my LAN only. I replaced the LAN 
switch with a new linksys 2024 with QOS and seemed to help but not fix 
the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, 
Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with 
an internal ip, one PAP2, one SPA3102 and 2 BT101. I know that bt's are 
cheap that are known for echo problem in the handset. I have one remote 
user that never has a problem. I have a remote test server at home 
connect via IAX with no problems, also a PAP2 with no problem. External 
faxing from the rest of the world via our voip provider is working 
great. One strange thing that I noticed is that we can not fax to our 
iaxmodem, ATA ---> iaxmodem, but works perfect ATA ---> rx_fax. Not sure 
why either.

Jonn

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Re: [asterisk-users] Issue Nortel CS2K/ISN08 to Asterisk Trixbox

2007-10-22 Thread Jonn Taylor 
[EMAIL PROTECTED] wrote:
> I have an trixbox(asterisk) software on a pc home edition.
> Origination is a Nortel ,model=CS2K,version=ISN08
> and my asterisk is doing termination.Nortel sent  calls
> to us ,Asterisk and they said that is sending call
> and i saw the trace as following:
>
> sip: [EMAIL PROTECTED] IP:5060 ;user phone
>
> but in my CDR i can view origination number but
> at destination i get 200 ,they said to me that they sent
> correct destination number in form of:
>
> CC+ area code +telephone number
>
>
> We do not have users or passwords.
>
> Nortel is given to me IP Signalling (1 IP) and media IP ,
> the scenario is IP to IP.
>
> In my side,asterisk ,i configured:
>
> 1. SIP trunk with :
>
> maximum channels=2
>
> Outgoing dial rule=cc+area +.
> Trunk peer
> ==
> allow=all
> context=from-internal
> host=Nortel Signalling IP
> port=5060
> type=peer
>
> Incoming setting on SIP trunks=nothing
>  register string=nothing
>
>
>
> 2.ZAP trunk
>
>outgoing dial rules
>===
>dial rules:cc+area code+.
>
>
> Please can you helping with this configuration or how i can
> configure ,the calls to come from Nortel to my asterisk?
>
> Nortel
> CS2k
> ISN08   .>SIP.>Asterisk trixbox
>
>
>
> Any help it will be higly appreciated
>
> Many thanks in advance,
>
> Tiberiu
>
>
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Change your type to friend not peer. This may help. Turn debug on in the 
cli console for the ip address that the calls are coming from, this way 
you can see if the info that they are sending is correct.

Jonn

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Re: [asterisk-users] Glare on Incoming Calls

2007-10-19 Thread Jonn Taylor 
Mojo with Horan & Company, LLC wrote:
> C F wrote:
>   
>> How on earth does this prevent Glare? Or even reduce it?
>>   
>> 
> I think he was providing his configuration in case there WAS a change he 
> could make to reduce it.
>
> The only thing we could do was an option because our incoming lines were 
> arranged in a hunt group.  We made sure that we dial out working down 
> the group.  So the phone company starts with line one, then line two, 
> etc., we start with line three, and then two...
>
> By using the Dial(ZAP/G1/blah) syntax.  The capital G searches the zap 
> channel group in reverse.  If you don't have a hunt group from the phone 
> company, this probably won't make a bit of difference to you.
>
> Moj
>
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They only way to eliminate a "glare" condition is to have your phone 
company convert you lines to ground start.

Jonn

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Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?

2007-10-12 Thread Jonn Taylor 
Olivier wrote:
>
> 2007/10/12, Jonn Taylor <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>>:
>
> > Would you then be able to port the given number to your new
> provider ?
> Yes, if you are in the US. Number port ability is one thing that ALL
> VIOP providers had to provide.
>
>
> Here too (France), number portability is mandatory but in facts, I 
> couldn't find any pure fax service provider complying with this.
> I think they bet on the fact they are not telco so they don't have to 
> comply with telco regulation.
> I could find fax services from telco but services are often poor or 
> neglected.
>
> That's fine you could find something up to your expectations : it 
> gives me hope I could find one in the future.
>
> Cheers
>
>
> 
>
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Olivier,

Check this out, might help.

http://www.voipproviderslist.com/country/voip-france/voip-providers-france/

Jonn

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Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?

2007-10-12 Thread Jonn Taylor 
Olivier wrote:
> John,
>
> For incoming fax numbers, did you port existing numbers or did you get 
> new numbers from bandwidth.com  ?
Both, we ported numbers and got new one's.
> If the later, what if you switch for another provider ?
I did alot of research before we went with bandwidth.com. They resell 
Level 3 service. If I would switch, I would verify that the provider 
that we switch to a) has very low latency b)has mutiple backup nodes. 
The other key is your internet provider, so long as they pass all TCP 
header info your good to go. We use Comcast Business service and get 
99.999% uptime.
> Would you then be able to port the given number to your new provider ?
Yes, if you are in the US. Number port ability is one thing that ALL 
VIOP providers had to provide.
>
> Regards
> 
>
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Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?

2007-10-11 Thread Jonn Taylor 
Mojo with Horan & Company, LLC wrote:
> Ex Vito wrote:
>   
>>   2. On the remaining locations "we have a problem"
>>   which I have been studying and trying to address...
>>   Faxing over IP.
>>   
>> 
> Could the 'remote' locations make do without a fax machine proper?  We 
> have sheet-fed pdf scanners here, drop the document in and hit the 
> button, and acrobat shows up; hit print, select the printer called 
> "Fax", hit OK, and type in a phone number.  Done.  The last bit (the 
> fake printer) is installed by "WinPrintHylaFax" [1] which is a windows 
> client that sends jobs over IP to a hylafax server.I'm not sure how 
> attached to a manual fax machine your users are, but mine sure were, and 
> this sheet-fed pdf scanner combined with winprinthylafax appeased them.
>
> Moj
>
>
> [1] http://winprinthylafax.sourceforge.net/
>
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We are faxing over SIP trunks from bandwidth.com and have 5 fax numbers 
all working without any problem. They are iaxmodem + hylafax. We can 
also send and receive faxes with tx_fax app. The big key is to have a 
bandwidth manager between your internet connection and your servers.

Jonn

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Re: [asterisk-users] server recommentation (unique requirements)

2007-08-28 Thread Jonn Taylor 
Stephen Kratzer wrote:
> Howdy. I've been having trouble finding a fairly modern server that meets the 
> following requirements:
> 
> - Molex power connectors (don't want to use the Digium FXS power supply)
> - 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC)
> - dual power supplies
> - preferably dual CPUs >= 1GHz
> - preferably rack-mountable (3-4RU)
> - CentOS-friendly
> 
> We'd also like to stay away from older HP servers. Any recommendations would 
> be greatly appreciated. Thanks.
> 
> Stephen Kratzer
> Network Engineer
> CTI Networks, Inc.
> 
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Dell PowerEdge 4400 should fit what you are asking for. The 2950 might also.

Jonn

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Re: [asterisk-users] Newbie Advice on Asterisk and Linux

2007-07-26 Thread Jonn Taylor 
Mark Burrows wrote:
> HI All,
> 
>  
> 
> I’m new to Asterisk and also to Linux.  I have a large IVR project that 
> I’m about to embark on.  I’m new to programming; new to Linux and new to 
> Asterisk.  I think I’m about to climb a steep learning curve.  I have an 
> existing IVR which is getting on for nine years old and is no longer 
> supported by my vendor.  I intend to replicate the system almost as is 
> and then add additional features and functions. 
> 
>  
> 
> I have been looking for a developer to put together my project and while 
> doing so have done lots of research and spoken to many people.  The 
> people who seem to understand my needs have recommended Asterisk.  For 
> the last couple of days I’ve been trying to look into Asterisk and learn 
> as much as I can; this has got me excited, motivated and a little 
> confused. Asterisk sounds like a great project and a great community.  I 
> think I have as much of an overview as I can.  Now I need to set up a 
> Linux system and get Asterisk running on it.
> 
>  
> 
> I’ve started to read the book Asterisk: The Future Of Telephony and 
> would like to now setup up a hobby computer to do some hands on 
> learning.  The book covers Red Hat Linux so I thought I’d look for a 
> ‘Red Had for Dummies’ book.  Even that got confusing. There’s Linux 
> Fedora, Enterprise Linux 4 and others.
> 
>  
> 
> Can someone suggest a starting point on learning Linux?
> 
>  
> 
> Thanks in advance,
> 
>  
> 
>  
> 
> Mark
> 
>  
> 
>  
> 
>  
> 
> 
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 
> 26/07/2007 9:56 AM
> 
> 
> 
> 
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http://www.trixbox.org

This is "one" of the many standard configs for aterisk. This uses CentOS 
4, Asterisk 1.2, FreePBX 2.2. You can setup a fully working system in 
about 30 min. Need help you can email me off list.

Jonn Taylor

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