[Asterisk-Users] Internet Telephony, net2phone

[Chris Mason]

As a newbie, can anyone advise me if Asterisk can route international calls
to a US based  service such as Net2Phone so we can take advantage of the
internet and save on calls? 

That would be my main reason for an Asterisk based PBX.

Chris Mason
[EMAIL PROTECTED]
Box 340, The Valley, Anguilla, British West Indies
Tel: 264 497 5670 Fax: 264 497 8463 Cell: 264 235 5670
http://www.anguillaguide.com/ The Anguilla Guide
Talk to me in real time:
Yahoo:netconcepts_anguilla
US Fax and Voicemail: (815)301-9759

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk PBX Billing

[Chris Mason]

I'm very interested in the same thing for a hotel system I would like to
implement. Anyone know if the country codes be tied to a pricing lookup
table?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi
Sent: Wednesday, July 02, 2003 5:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Billing


Hi

I would like to use the Asterisk PBX as part of a phone shop system instead 
of the usual PBX plus PC. How can I do the the billing in a way that is 
convinient to the phone shop attendant?

Regards

Shepherd

_
Add photos to your messages with MSN 8. Get 2 months FREE*. 
http://join.msn.com/?page=features/featuredemail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing

[Chris Mason]

That's all I would need, it would be easy enough to work out the cost after
that.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro
Sent: Wednesday, July 02, 2003 10:06 AM
To: Scott Stingel
Cc: [EMAIL PROTECTED]
Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing


i think that the problem could be something more easy:

it is possible inside asterisk to log all che calls of all the users and
know the timing and the number called for each call? if it is possible to do
that, could be possible to make a program that takes this files and generate
the costs reading the log informations...

so for me the real question is: there is a log of all the phone call that
are made by asterisk?

Angelo



Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS> Shepherd-

SS> Having designed one of these in the past (in a higher level voice 
SS> environment), I can tell you that this is not a small undertaking.  
SS> It's at least as much an SQL job as a voice task.

SS> Usually the way to accomplish this is to establish more-or-less a 
SS> pre-paid phone card system, where the shop prepays an overall amount 
SS> for international calling access.  Then you have to time each call 
SS> as it is occurring, debiting each account, and the master account, 
SS> in real-time. This can be a bit complex when you have 20 or 30 calls 
SS> going at one time.  You have to cut them off promptly when the money 
SS> runs out (big problem).  And you have to provide call detail and 
SS> charges to them at the end of each call, using their own retail 
SS> tariff.

SS> To add to the complexity, each country has a different tariff from 
SS> the long distance carrier, and within the country, major cities 
SS> often have special rates per minute.  Mobiles have a different 
SS> tariff too.  Phone card platforms usually include a least-cost 
SS> routing system which chooses a carrier real time based on the call.  
SS> Tariffs change weekly and must be updated in the system.

SS> Anyway, I'm just scratching the surface!  I'll write more when I 
SS> can!

SS> Cheers
SS> Scott Stingel


SS> Scott M. Stingel
SS> Emerging Voice Technology Inc.

SS> Email:  [EMAIL PROTECTED]    
SS> URL:www.evtmedia.com    



>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> shepherd fungayi
>> Sent: Wednesday, July 02, 2003 10:49 AM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Asterisk PBX Billing
>> 
>> 
>> Hi
>> 
>> I would like to use the Asterisk PBX as part of a phone shop
>> system instead 
>> of the usual PBX plus PC. How can I do the the billing in a 
>> way that is 
>> convinient to the phone shop attendant?
>> 
>> Regards
>> 
>> Shepherd
>> 
>> _
>> Add photos to your messages with MSN 8. Get 2 months FREE*.
>> http://join.msn.com/?page=features/featuredemail
>> 
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED] 
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 



SS> ___
SS> Asterisk-Users mailing list
SS> [EMAIL PROTECTED] 
SS> http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Angelo Sampietro
IT Manager
ARC Interactive

"After a certain high level of technical skill is achieved, 
Science and art tend to coalesce in esthetics, plasticity, and form. 
The greatest scientists are always artists as well."
 
Albert Einstein 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?

[Chris Mason]

hank wrote:


there is no list for [EMAIL PROTECTED]
and why aint it aloud on this list its asterisk related aint it?
chill out guys


AAH is an abstraction layer for Asterisk, and the issues that relate to 
it and not Asterisk belong on it's own list. If there is no list, which 
would surprise me, use the forums on sourceforge.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Colocation/Telehousing

[Chris Mason]

Sahil Gupta wrote:


Hi,
Is there anybody on the list that recommends anyone for 
colocation/telehousing in the US?


I'm after 2 Servers to be hosted in the US, preferably on the west coast.


Talk to [EMAIL PROTECTED], he can give you a server with a real TI 
terminated to PSTN, and at excellent rates.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Users handbook

[Chris Mason]

At the most recent project I completed I have to post a intranet web 
page with instructions on using the system and phones. Asterisk is 1.07 
stable and the phones are Polycom IP300, IP500, and IP600.
Has anyone done an Astersik users guide? Something non-technical but 
covering most of the features an office worker would use.
If nothing exists, should we develop this as a documentation project? 
After all, the greatest software is little use if the users never hear 
about the features.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax

[Chris Mason]

Cullin J. Wible wrote:

We are unable to call certain 800 numbers through Teliax but I thought 
I would post this here and see if anyone else had the same problem 
with either Teliax or other carriers.


The 800 numbers causing problems pick-up the call right away and are 
in the US - American Airlines (8004337300) and Staples (800-378-2753) 
- we can call many other 800 numbers just fine.


My users have reported the same problem with AA, we also use Teliax. I 
coul care less about Staples but American Airlines is the airline that 
serves this destination, so it is important to us.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom configs?

[Chris Mason]

Ted Serreyn wrote:


Ditto, I only have a couple of the polycomm phones spent the better part of
1 day figuring out how to get them configured properly.


 

I've had a lot of response like this which leads me to think there is a 
need for some documentation on the subject, I know I found that lacking 
when I was setting up the first phones. I have now done lots of 
variations with both DHCP provisioned and stand alone configured ftp 
settings with multiple registrations, public NTP, all picking up config 
files from a central FTP server. I have also done web configured phones 
but that sucks.
What does the list suggest as the proper place to document installation? 
I would be happy to expand the voip-info wiki on polycoms, and develop 
the script to be mroe generic. Right now it is purely for my own use, 
there is no error checking, no validation, no help, documentation or 
options. Please let me know what the script should do.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom configs?

[Chris Mason]

Gonzalo Gonzalez wrote:


I have tried for weeks now trying to config my first IP301 and the only way
I get the phone registered to asterisk is if I set the server and user info
on the web interface.  I have tried all different settings with the boot
server?  What I am doing wrong?
 

I am copying you the script, make sure you have setup the DNS to point 
ftp to the right server, set the PlcmSPIp users's password on the server 
and on the phone, and that the phone downloads the config file (watch 
/var/log/messages and /var/log/xferlog to determine problems).
Also, very important, before you run the script, edit the variables in 
it and also:

tar cfz asterisk-configs.tgz /etc/asterisk

Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicemail management

[Chris Mason]

C F wrote:


Just run somthing like this:
rm -R /var/spool/asterisk/vm/default/1xx/* (I think this should do,
otherwise something similiar will).

 

Yeah, I'm sittng around waiting for guests to check out! No, this is a 
job for php and an authenticated web page.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom IP600 - Flashing clock and date?

[Chris Mason]

Don't put any SNTP entry in the MAC-phone.cfg file.

Make sure the ntp server is configured to allow external access.

restrict 192.168.0.0 mask 255.255.255.0 notrust nomodify notrap

Config /etc/dhcpd.conf file

#
# 192.168.0.0/255.255.255.0 Scope Settings
#
subnet 192.168.0.0 netmask 255.255.255.0 {

   # Range of DHCP assigned addresses for this scope
   range   192.168.0.100 192.168.0.199;
   # 1 day
   default-lease-time  86400;
   # 2 days
   max-lease-time  172800;

   # Configure the client's default Gateway:
   option subnet-mask  255.255.255.0;
   option broadcast-address192.168.0.255;
   option routers  192.168.0.1;
   
   # Configure the client's DNS settings:

   option domain-name  "mason.home";
   option domain-name-servers  192.168.0.3;
   option ntp-servers  192.168.0.3;
   option routers  192.168.0.1;
   option tftp-server-name "192.168.0.1";
   option time-offset  -14400;
  
   # If you want the client to be configured to also use

   # a WINS server:
   option netbios-name-servers 192.168.0.3;
   option netbios-node-type8;
  
}  
~  


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] what is the problem with gmail and the list.

[Chris Mason]

--- Time Bandit <[EMAIL PROTECTED]> wrote:

> > I have not been receiving mail from the list 29th
> July, what is the problem
> > with gmail and the list. 
> No problem here.
> 
Mine stopped on the same data, July 29. I had to
subscribe as a new account to get mail from the list. 
I checked the log of my mail server, no mail seen
since than.

Chris


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] How to use multiple VOIP provider trunks

[Chris Mason]

I have been able to setup three different providers successfully, but only
one at a time. I would like to have all active in a fail over configuration
so that one failing would not be noticed by the users. I know it's probably
easy to configure but I have not been able to find out how. Can anyone give
me an example?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] How to use multiple VOIP provider trunks

[Chris Mason]

So, can I take it that most admins are using one provider or doing the
switch over manually when there is a problem? I have been testing voipjet
and it has good quality, how has the reliability been?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Start on system restart

[Chris Mason]

How should I get asterisk to start automatically on system restart?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Start on system restart

[Chris Mason]

Thanks, worked beautifully.

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fail over

[Chris Mason]

For all my PBX installations I want to have Fail Over on the main incoming
PSTN line so that a power outage does not leave the offices stranded. Is
there any commercial solution to this? I would rather a finished product
than a home soldering project.

Chris Mason
[EMAIL PROTECTED]
Box 340, The Valley, Anguilla, British West Indies
Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483   
Fax: (264) 497-8463 - US Fax (815)301-9759
Yahoo IM: [EMAIL PROTECTED]
Skype ID: netconcepts

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Fail over

[Chris Mason]

No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.

 What I need is a little box that diverts calls if the PBX goes down.

Chris Mason
[EMAIL PROTECTED]
Box 340, The Valley, Anguilla, British West Indies
Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483   
Fax: (264) 497-8463 - US Fax (815)301-9759
Yahoo IM: [EMAIL PROTECTED]
Skype ID: netconcepts

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthew Marlowe
> Sent: Tuesday, March 29, 2005 9:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Fail over
> 
> 
> There's many solutions.. One being www.voiceguard.com I think 
> might be what 
> you want.
> 
> - Original Message - 
> From: "Chris Mason" <[EMAIL PROTECTED]>
> To: 
> Sent: Tuesday, March 29, 2005 8:01 AM
> Subject: [Asterisk-Users] Fail over
> 
> 
> > For all my PBX installations I want to have Fail Over on 
> the main incoming
> > PSTN line so that a power outage does not leave the offices 
> stranded. Is
> > there any commercial solution to this? I would rather a 
> finished product
> > than a home soldering project.
> >
> > Chris Mason
> > [EMAIL PROTECTED]
> > Box 340, The Valley, Anguilla, British West Indies
> > Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483
> > Fax: (264) 497-8463 - US Fax (815)301-9759
> > Yahoo IM: [EMAIL PROTECTED]
> > Skype ID: netconcepts
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users 
> 
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Fail over

[Chris Mason]

 
> Some of their products are programmable too, where you can 
> send TCP messages to 
> initiate the switching process. Check out their website for 
> more products.
> 

That's perfect, because I use a Nagios monitoring system that can tell if
the Asterisk system is running and tell the fail-over switch to switch if it
isn't. I'm not sure how to monitor Asterisk yet but it looks like this will
do it:
http://megaglobal.net/docs/asterisk/monitor_pbx.pl

Together with that I would monitor disk space, cpu load, http and ping,
should make sure everything is working well.

The only other issue is power failure but with a large UPS system I don't
expect that to be an issue.

Chris

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Combatting echo in VOIP

[Chris Mason]

I am currently evaluating Asterisk as a replacement PBX and currently
testing the VOIP trunks with voipjet as the provider, connecting using IAX.
I have been making calls all over the world to test the quality of this
provider.
Although I hear perfect audio, the receiving party hears a little echo on my
voice. Is there anything I can do about this?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problem with livevoip dial out

[Chris Mason]

I am starting to use livevoip but when I configure they way
they suggest, I see errors. 

 

[livevoip]

exten
=>_51NXXNXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})

exten => _51NXXNXX,2,Hangup

 

I want the users to dial 5 to get a livevoip trunk.

 

Here’s the error message:

 

    -- Executing
Dial("IAX2/[EMAIL PROTECTED]:4569-6", "1000|15") in
new stack

Mar 31 22:31:07 WARNING[27589]: app_dial.c:920
dial_exec_full: Dial argument takes format
(technology1/[device:]number1&technology2/[device:]number2...|optional
timeout)

 

 

Chris Mason

 

 

 

 

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Sending faxes and call accounting

[Chris Mason]

>I don't understand you're confidentiality arguement. If asterisk is
switching the call, it /can/ save a copy of the transmission.

Of course, we know that. But the perception is that the fax machine is
private, so that's what the clients want.

>None the less, you should be able to switch a fax call just like a voice
call.
That's not what I am reading, but I will set up a test.
I would like to run hylafax for another installation, my own office, but I
am not clear on how this is achieved. I already run hylafax on another
machine, but what I cannot figure is, if I run it on Asterisk do I need the
modem, or does it receive on the FXO port the line is connected to?

cheers,
glenn


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Digium TDM400 Failover on Power Loss

[Chris Mason]

This site has what you need and some other products that might also work
http://www.gkinc.com/OverviewUT8.htm

They will sell direct to you. I haven't purchased yet but I intent to use
them for the same reason. You could trigger off the computers power supply
so turning off the server flips the lines over to standby handsets.

Chris

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Low volume in recorded messages

[Chris Mason]

My voice messages emailed to me have poor volume, to the
point where I can barely hear them. I have confirmed by loading them in
Audition that the level is poor. This happens on PSTN and VOIP calls. Recording
my own voice messages using the record application results in very low levels
also.

 

Is there any way to adjust the levels?

 

 

Chris Mason

 

 

 

 

 

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] unlimited iax termination

[Chris Mason]

Serves you right for offering a bait and switch deal. If you are selling
"unlimited" that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a "1000 minutes for $10" plan? I guess you are
afraid someone will then offer an "unlimited" plan and take all the
business! So you all offer unlimited, even though you can't deliver it and
hide the real details in the fine print. So much for truth in marketing.
There's laws to protect us from this kind of marketing, it's a shame they
aren't used more often.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 
> Beware of internet cafes/call shops.  They love these plans.  They sign
> up from provider to provider using different credit cards.  It takes a
> while for them to catch them.  By the time the jig is up they have
> milked several hundred thousands minutes on a single account.  Try to
> make a profit on that at $19.95 :)  It has happened to us and it will
> happen to you.  Unless you put a daily limit of X amount of minutes this
> is hard to catch in under 30 days.
> 
> --
> Andres
> Network Admin
> http://www.telesip.net
> 
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Problem with Livevoip incoming context

[Chris Mason]

Done all that, still doesn’t work.

I do have outgoing and incoming, just can’t
get the incoming to come through the livevoip context.

Thanks



Chris Mason
US Number: (646)722-0001 US
Fax (815)301-9759
Skype: netconcepts
  













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Thursday, April 14, 2005
5:50 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problem with Livevoip incoming context



 

Should have in iax.conf.

 

;This registers you to them

register=:@64.34.59.73

 

;THis context serves to ID incoming, if
you ahve a DID it shoudl come here

[livevoip]

type=user

secret=mySecret

host=64.34.59.73

callerid="Livevoip IAX User"

context=livevoip-in

 

;This one is your outgoing...

[ToLiveVoIP]
username=username
type=peer
secret=YourSecret
host=64.34.59.73



 





 





As long as your Dial Plan refrerences these correctly, you
should get both in and out with incoming registered to your livevoip.





 





Wiley





 





 





 





 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Thursday, April 14, 2005
2:39 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problem
with Livevoip incoming context

I have a newly provisioned livevoip account which registers
OK but the incoming calls are not being authenticated as livevoip and only work
as the guest context:

 

 

[livevoip]

type=user

secret=mySecret

host=64.34.59.73

callerid="Livevoip IAX User"

context=livevoip-in

 

[guest]

type=user

callerid="Guest IAX User"

context=guest-iax-in

 

 

Any ideas?

 



Chris Mason

www.anguillaguide.com



 








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Problem with Livevoip incoming context

[Chris Mason]

The connection does not authenticate. What difference does the dialplan
make, this happens way before the dialplan.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Moody
> Sent: Friday, April 15, 2005 7:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Problem with Livevoip incoming context
> 
> Can you post your dialplan?
> 
> If you are using the extension 's' change it to the actual DID number.
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Cisco's description of echo

[Chris Mason]

>Another common problem that causes echo in networks is not setting your
loss plan correctly.    You need to be sure that you aren't coming in too
hot at any of your analog interfaces.   In general you should see a signal
between -20dbm and -12dbm when someone is talking on the line.   If it is
significantly hotter then you run the chance of having a larger reflected
signal resulting in echo.   I typically try padding down analog levels by
3dB at a time to see if echo is reduced.   


How do you measure the amplitude of a pstn line? As an audio engineer in a
previous life, I would love to be able to send standard level tones down a
pstn line and measure the amplitude at my end, then adjust the input gain
accurately. Is there a way to do this?


Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
  
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Call Recording via monitor

[Chris Mason]

> Unfortunately the otherwise excellent Areski stat tool doesn't seem
> to include the unique ID function and thus I can't pull a file back
> directly from that tool
> 
> Anyone fancy some development activity?

If it works out I will sponsor some development in this area, we need this
interface. If anyone else can chip in please let me know.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Digium MOH

[Chris Mason]

Why not?

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Satellite Providers

[Chris Mason]

 

The delay in the air is minor. Radio travels very fast through the air. Almost at the speed of light. 

 

It may travel very fast but it’s
also a very long way, 22,000 miles up, then 22,000 miles down, then the same
all over again. The latency for satellite is about 500ms round trip, that’s
a lot. It’s on the very edge of what’s possible for VOIP.

I looked at Satellite and decided it was
not competitive.

 

Chris








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] IAX registration refused

[Chris Mason]

What does trunk=yes do?

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Polycom configuration

[Chris Mason]

How do you configure your Polycom phones? Is it enough to configure one line
appearance? Or is there a way to configure a roll over?



Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Road Warrior phone config

[Chris Mason]

No, he wants a desk phone. He could take the phone with him but he doesn't
want to. I like the example from Robert, I'm going to try that.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andres Paglayan
> Sent: Sunday, May 15, 2005 6:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Road Warrior phone config
> 
> question about this thread,
> would a wi-fi voip phone work for this guy?
> meaning, he takes it to wherever he goes and it gets registered wherever
> it as wireless access.
> is that theoretically correct?
> 
> >
> >
> >
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] POE hub

[Chris Mason]

I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.

Chris Mason


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] POE hub

[Chris Mason]

> Lol - yeh and at $1300 I prefer some power plugs.

That's how I feel

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Help with extensions - can't dial 700

[Chris Mason]

I have been working on integrating some FXS ports into my dial plan
delivered via a channel bank and testing with an analog handset.

The receptionist is on Extension 700. All other SIP phones are 7XX.
>From a SIP phone I can dial 700 and all other extensions.
>From the analog handset I can dial any other extension but not the 700
number. Weird? Yep.
The CLI does not show any dialing when I dial 700.

Extensions.conf

[office]
exten => 700,1,Dial(SIP/700,20)
exten => 700,2,VoiceMail,u700
exten => 700,102,VoiceMail,b700

exten => 701,1,Dial(SIP/701,20)
exten => 701,2,VoiceMail,u701
exten => 701,102,VoiceMail,b701

exten => 702,1,Dial(SIP/702,20)
exten => 702,2,VoiceMail,u702
exten => 702,102,VoiceMail,b702

exten => 703,1,Dial(SIP/703,20)
exten => 703,2,VoiceMail,u703
exten => 703,102,VoiceMail,b703

...

[zap-in]
exten => s,1,NoOp(Starting Zap-in)
include => office

==
zapata.conf
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

context=zap-in
group=1
signalling=fxo_ks
faxdetect=incoming
channel => 1-12



Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help with extensions - can't dial 700

[Chris Mason]

Thanks, I removed that and will test. I don't have an analog extension here,
I am testing using SIP remotely, will have to go to the resort to test.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] VoipSupply.com

[Chris Mason]

"I have gotten"
What language is that?

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Analog Lines

[Chris Mason]

I can put you in touch with a used equipment dealer that sells 24 port
Adtran channel banks for less than $500. I got one and there was one module
with 2 UBRt1E channels, he fedexed the 4 x FXO module to me in Anguilla
right away and didn't require the old one back. Great service andreliable
equipment, should make a channel bank the best choice for your installation.

Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
 
> >
> > I have looked at going with a T1 interface to Channel Bank, but that
> > just seems like a very expensive way to solve this problem.  ($1500 -
> > $2000 ).
> >
> > Any suggestions?
> >
> > Sean


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] No recorded messages

[Chris Mason]

I have installed my first Asterisk implementation using the [EMAIL PROTECTED]
ISO. I am using the SJPhone software. Using the setup page, I have been able
to configure two extensions. Whne I dial from one to the other, the other
does not answer even though it is registered. Watching the log in the CLI, I
can see that recorded messages are being played;:
  == No one is available to answer at this time
  dialparties.agi: Dial return value was 0 and dialstring was SIP/203|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Wait("SIP/202-ed3d", "1") in new stack
-- Executing VoiceMail("SIP/202-ed3d", "u203") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/203/INBOX/msg0005 format: wav49,
0x93b4280
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/203/INBOX/msg0005 format: wav,
0x941e3f0
-- Recording automatically stopped after a silence of 5 seconds
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')


I do not hear these messages. I can leave a voice message and the other
extension can play it, so some of it is working fine.Any ideas hwy
1: No recorded messages
2: Extensions do not answer. It looks like it should when I observe the log.

dialparties.agi: Caller ID name is 'Chris Mason' number is '202'
--  dialparties.agi: Added extension 203 to extension map
--  dialparties.agi: Extension 203 cf is disabled
--  dialparties.agi: Extension 203 do not disturb is disabled
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
  dialparties.agi: Extension 203 has call waiting disabled
--  dialparties.agi: DbSet CALLTRACE/203 to 202
  dialparties.agi: About to execute Dial(SIP/203|15|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/203|15|tr)
-- Called 203
  == No one is available to answer at this time

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Backup for linux/asterisk

[Chris Mason]

I gave up on tape as being a nightmare to maintain, I now back all my
servers and workstaions using backuppc. One linux server with a 5 device
RAID can easily backup 100 workstatons and several servers beacuase of the
pooling system used. For a smaller situation I would use 2 disks in RAID1
(mirror).

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with DTMF on IVRs

[Chris Mason]

I tried DTMFmode=auto and it did not help. Any further ideas?

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fail over using CHANAVAIL

[Chris Mason]

I am trying to construct a macro for long distance dialling. I have two 
internet feeds, I have all routes including Teliax on Internet A and a 
static route to Voxee on Internet B. I thought I could use the dialplan 
entry below which uses the ChanIsAvail() command to check the 
connection, but this returns the provider but not the username, so I 
don't understand how to use this for real applications to determine IAX2 
availability. The only way I can see to use it is to only specify one 
channel and test it, jumping to n+101 if it isn't.


[globals]
[EMAIL PROTECTED]
[EMAIL PROTECTED]

[macro-longdistance]
;
; Standard extension macro:
;   ${ARG1} - Number to dial
;
exten => s,1,SetCallerID("NetConcept"<1234567890>|a)
exten => s,2,ChanIsAvail(IAX2/${TELIAX}&IAX2/${VOXEE})
exten => s,2,Read(${AVAILCHAN})
exten => s,3,Cut(C=AVAILCHAN,,1)
exten => s,4,NoOp(AVAILCHAN= ${C})
exten => s,5,Dial(${C}/${ARG1},60,tr)

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] low sound

[Chris Mason]

jonny hashem wrote:


my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change? 





Could you be more vague?
Try giving us hardware, relevant config sections,  type of CO lines, 
someting to work with.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)

[Chris Mason]

JP Carballo wrote:


Chris Mason (Lists) wrote:



 

Overall, I'm happy. It has sturdy construction, standard features, 
and most of all works just fine with *.


Did you find any noise, hum or gating on the FXO ports on incoming 
calls?



Nope.  Crystal clear calls.

That's PSTN -> VG-400 -> Phone and PSTN -> VG-400 -> * -> VG-400 -> Phone


Any idea where I can ge them?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...

[Chris Mason]

Shaun Bolling wrote:

Jonathan, did you have any problem getting your polycom 301 to work 
with asterisk. I purchase two of them for testing. I have been trying 
for two days now to get them to call one another, with no luck. My 
software phones work fine. In my asterisk log I get a error message 
like "Failed to authenticate user". Did you have this problem?


Send me the info and I will write you a config file for one phone:
MAC of phone
serverip
username (for ftp account)
password (for ftp account)
Extension number
codec preference

I'm presuming you are using DHCP to allocate TFTP setting, SNTP server, 
phone IP, Gateway.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Channel bank timing

[Chris Mason]

Andrew Kohlsmith wrote:

Tell the Channel Banks clock to the line, and have the Sangoma card NOT sync 
to anything (i.e. the A104 is the master, the channel banks the slaves).
 


I set the card up so that
Port1
TE_CLOCK= MASTER
TE_REF_CLOCK= 0
Port2
TE_CLOCK = NORMAL
TE_REF_CLOCK = 1

which should make Port 2 take it's timing from Port 1 and Port 1 take 
it's timing from the onboard clock.


Basically clocking works this way:  Each end of a T1 sends data generated by 
an on-board clock.  These two clocks (one at each side) needs to be in 
perfect sync with each other or you get frame slips and other nasties.  The 
solution is to have one of these clocks lock or synchronize to the far side.  
This is know by several names, among them "line clock", "recovered clock", 
"slave clock", etc.  The side that is not trying to synchronize is also known 
my several names... "master clock", "internal clock", etc.
 

On the 600 I set it to Timing = Network, but on the 750 I can't figure 
out which one of these it should be.

LOOP
LOCAL
EXTERNAL

On the 600, the manual says:
"The selected clock option always designates the clock source for 
transmission. Clocking necessary for receiving data is always recovered 
from incoming data."


I think the 600 manual also gives me the answer for the 750:
Network Timing - The network is the source of timing. The received data 
clocking is looped back to the network, where it is used to determine 
the transmission timing. This option is also referred to as loop timed 
as the transmission clock is derived from the received clock.


So for the 750, loop would be the same thing.

So, as far as I can tell, everything is set correctly. Which is a 
problem because it does not sound right.


 



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] name that vendor...

[Chris Mason]

I've got one, they suck pretty bad.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Satellite receiver over IP

[Chris Mason]

I need my satellite receivers to call home to avoid problems with the 
service. I have hooked them up through an IAXY and tried a SPA2002 set 
to G711 and made sure the transport is 711 all the way. However, it does 
not work at all, the receivers cannot make the connection work.

Has anyone made this work?

Chris Mason
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] DNS Server Failure wreaks havoc

[Chris Mason]

Eric "ManxPower" Wieling wrote:


Brian Capouch wrote:

I don't think this is a new issue--I've seen it talked about on the 
list before.  I don't know if I've ever seen anyone post a fix.


My DNS server went out last night in a horrendous storm when an 
upstream link went down.  The madness is that the behavior of the 
whole server, including the part that's handling my POTS lines, gets 
wigged out on a DNS failure, making the whole system unusable.  I 
have two questions; being able to solve either would be wonderful:



Asterisk is horrible at handleing DNS failures.  Don't use DNS with 
Asterisk.


I have found Asterisk is terrible with any kind of internet outage. IAX 
stops trying to register if the internet goes down for a few minutes and 
the customer looses long distance calling until a tech resets the PBX. I 
had to setup a cron job to reload asterisk every hour to get any kind of 
reliability.
I also found the same problem with local calls, when I lose internet the 
locals calls go out also. How can that make any sense?


Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hi Honies! I'm home!

[Chris Mason]

Your dinner's in the oven.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] New ncurses Asterisk Manager Interface

[Chris Mason]

Sig Lange wrote:


The projects real home will be assman.sf.net .
Feedback is welcome as well as requested features.


ROTFL - you have go to be kidding.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom 601 question

[Chris Mason]

Kevin Smith wrote:


It is making me lean that way, because other phones (same settings) 
are using the AC adapters in another office. The ones on the adapter 
have not been having this problem, but they don't use the phone much 
so they may have never noticed if it did.


If you go into the ftp site for the polycom phone configs, cd to logs 
and run "ls -lrt", you will see which phones have rebooted in the order 
they rebooted.
Get the mac address of the stand alone power supply phones and see when 
they last rebooted.


Also, add a power supply to the ones that are rebooting, see if that 
stops the problem.


--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Recommended FXO device

[Chris Mason]

I have a client's installation that requires 4 lines PSTN interface only 
so I am looking at 4 port FXO units. What works well with Asterisk and 
is not exorbitant to purchase? Would a Sangoma remora be better?


Chris Mason


--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Calling on all Polycom Experts

[Chris Mason (Lists)]

>Could not load time from 0.0.0.0(0.0.0.0).

No dns server to resolve SNTP address

Other than that I don't see a problem. You need to look at the
logs/MAC-app.log for more clues.



What I find useful is to download a free XML editor and load the config file
into it, that will test for xml syntax errors you may not see by eye.
Also, take everything out except the following and build from there.

 
  
   
  


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Includes include the includes?

[Chris Mason (Lists)]

I am grouping my extensions by building like so:
1XX  is Building 1
2XX  is Building 2
7XX  is Office

[Office] extensions has the following includes
7xx
Include => Local
Include => International
Include => Building1
Include => Building2

[Building1] has 
1xx
Include => Office
Include => Building2
Include => Local

I done't want building1 to access international, but does it inherit that
include through including the office context?


Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Do includes include the includes

[Chris Mason (Lists)]

I am grouping my extensions by building like so:

1XX  is Building 1
2XX  is Building 2
7XX  is Office

[Office] extensions has the following includes 
7xx 

Include => Local 
Include => International 
Include => Building1 
Include => Building2

[Building1] has
1xx
Include => Office
Include => Building2
Include => Local

I don't want building1 to access international, but does it inherit that
include through including the office context? If it does, how can I
structure a dialplan so that each building can call each other but building1
does not have international?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fall back dialing

[Chris Mason (Lists)]

We have ServerA that connects to ServerB to dial long distance via an IAX2
trunk. I have setup an international dialing plan so that there is a backup
route via pstn if the IAX channel is down.

exten => _1NXXNXX,1,Dial,IAX2/${SERVERB}/${EXTEN},60)
exten => _1NXXNXX,2,Dial(Zap/g2/${EXTEN},70)
exten => _1NXXNXX,3,Macro(fastbusy)
exten => _1NXXNXX,4,hangup
exten => _1NXXNXX,102,Dial(Zap/g2/${EXTEN},70)
exten => _1NXXNXX,103,Macro(fastbusy)
exten => _1NXXNXX,104,hangup


Would it be better to use

exten => _1NXXNXX,102,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?i,1:103)


Any other improvments? We want to make is transparent to the users.

Chris Mason
NetConcepts
Int:  (305) 704-7249 Fax: (815)301-9759 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Do includes include the includes

[Chris Mason (Lists)]

I am grouping my extensions by building like so:

1XX  is Building 1
2XX  is Building 2
7XX  is Office

[Office] extensions has the following includes 7xx 

Include => Local
Include => International
Include => Building1
Include => Building2

[Building1] has
1xx
Include => Office
Include => Building2
Include => Local

I don't want building1 to access international, but does it inherit that
include through including the office context? If it does, how can I
structure a dialplan so that each building can call each other but building1
does not have international?

Chris Mason

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Includes include the includes?

[Chris Mason (Lists)]

First, let me apologize for the multiple posts - my procmail recipe had a
bug that hid most mail form the list for a day.

The inheritance of includes creates a problem for me. I want to group the
extensions, not put them all in default to control access to features. So
[office] extensions should have the include => longdistance but  [building1]
should not.

However, how can [building1] then dial office?


Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759  

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tarpo, Louie
> Sent: Wednesday, June 15, 2005 9:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Includes include the includes?
> 
> Yes it does.  You want something like this...
> 
> [office]
> include => default
> include => local
> include => international
> 
> [building1]
> include => default
> include => local
> 
> [default]
> exten => 700,1,Dial(SIP/${EXTEN})
> exten => 100,1,Dial(SIP/${EXTEN})
> exten => 200,2,Dial(SIP/${EXTEN})
> ;and so on for your other extensions
> 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Zombie?

[Chris Mason (Lists)]

What does the Zombie mean in this line from the CLI?

== Spawn extension (daymenu, s, 1) exited non-zero on 'SIP/700-fe13'

Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Garbled one-way audio only with ulaw

[Chris Mason (Lists)]

[EMAIL PROTECTED] wrote:

For some reason a couple weeks ago users began experiencing garbled audio 
in one direction when dialing out via our VoIP provider. 

Play with the jitter buffer, I'll bet this is your problem. I had 
exactly the same problem with a cable ISP. Also, watch for strange routing.


Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] AGI to monitor conenction quality

[Chris Mason (Lists)]

I need an AGI to monitor the quality of two connections and return a 
yes/no based on packet loss, connectivy, provider being there, so I can 
rollover the dial plan and dial the next available method. We have two 
internet connections, two providers, and PSTN for backup.


My main concern is to make sure the call gets connected one way or the 
other, cost being secondary but also important. I am not looking to 
determine the quality of the VOIP provider, just the network.

Has anyone got some code?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] mini itx

[Chris Mason (Lists)]

I have a 3 GHz Mini ITX with 1 GB ram, I don't think it would be much of 
a feat to run Asterisk on it. In fact, that;'s my demo system, small, 
light, looks great and very fast. The only downside is the single pci 
slot, with a T1 card in there there no room for an additinal NIC.


jltaylor wrote:


I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 




--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] mini itx

[Chris Mason (Lists)]

jltaylor wrote:


It may not be enough horsepower...
I'm looking for a "black box", with a PCI slot to put in a telco closet.
Needs to be able to take the 4 port T1 card (pci slot) and do g729 for 50-60
calls.

Any suggestions?

 


A shuttle with a 3GHz CPU will do it without a problem.

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk server with remote monitoringcapabilities

[Chris Mason (Lists)]

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.)
   

I have never had a pbx lock up. I suggest you change your hardware. I 
would think your problem is RAM, as Asterisk is not hard on the drives 
or other hardware.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Teliax Problems

[Chris Mason (Lists)]

An ethereal trace indicates the IP address is active, but it is not
responding to iax packets (registration). So, either their asterisk
app has failed or they have folded their tent as well.


 

I am sure it's just a crashed server, wait an hour and let the support 
people deal with it.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Server hanging

[Chris Mason (Lists)]

I have a dedicated server running Asterisk to connect a geographically 
seperated sales force of seven people. There is no Asterisk hardware in 
the server, the connections only SIP and IAX.
The OS ia Red Hat Enterprise Linux ES release 3 (Taroon Update 5), the 
phones are Polycom IP500 and IP600s.


In the past three days, the server has stopped connecting the phones and 
the phones are not registerd. However, Asterisk is running and there is 
nothing in the logs, I can access the CLI and when I run "sip show 
peers" shows all the phones as registered. I have several Asterisk 
installations and none of the others show this behavior. Stopping and 
restarting Asterisk clears the problem.


I'm at a loss to explain it to the client, any ideas on how to solve this?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Can't build cdr_addon_mysql.

[Chris Mason (Lists)]

I have been unable to build cdr_addon_mysql from asterisk-addons-1.09. 
Could it be a mysql4 issue


[EMAIL PROTECTED] asterisk-addons-1.0.9]# make cdr_addon_mysql
cc -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql -c -o 
cdr_addon_mysql.o cdr_addon_mysql.c

cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory

I have MySQL devel

# ls /usr/include/mysql/
total 404
drwxr-xr-x2 root root 4096 Jun 30 00:21 .
drwxr-xr-x   67 root root 8192 Jun 30 00:21 ..
-rw-r--r--1 root root 4544 May 13 06:14 chardefs.h
-rw-r--r--1 root root 3428 May 13 06:14 errmsg.h
-rw-r--r--1 root root 9472 May 13 06:14 history.h
-rw-r--r--1 root root 6774 May 13 06:14 keycache.h
-rw-r--r--1 root root 3529 May 13 06:14 keymaps.h
-rw-r--r--1 root root17851 May 13 06:14 m_ctype.h
-rw-r--r--1 root root 8085 May 13 06:14 m_string.h
-rw-r--r--1 root root 1858 May 13 06:14 my_alloc.h
-rw-r--r--1 root root28423 May 13 06:14 my_config.h
-rw-r--r--1 root root 3463 May 13 06:14 my_dbug.h
-rw-r--r--1 root root 3509 May 13 06:14 my_dir.h
-rw-r--r--1 root root 3000 May 13 06:14 my_getopt.h
-rw-r--r--1 root root38891 May 13 06:14 my_global.h
-rw-r--r--1 root root 1510 May 13 06:14 my_list.h
-rw-r--r--1 root root 3588 May 13 06:14 my_net.h
-rw-r--r--1 root root 1245 May 13 06:14 my_no_pthread.h
-rw-r--r--1 root root24524 May 13 06:14 my_pthread.h
-rw-r--r--1 root root 1784 May 13 06:14 my_semaphore.h
-rw-r--r--1 root root15028 May 13 06:14 mysql_com.h
-rw-r--r--1 root root11315 May 13 06:14 mysqld_error.h
-rw-r--r--1 root root 1241 May 13 06:14 mysql_embed.h
-rw-r--r--1 root root27544 May 13 06:14 mysql.h
-rw-r--r--1 root root 2184 May 13 06:14 mysql_time.h
-rw-r--r--1 root root  821 May 13 06:14 mysql_version.h
-rw-r--r--1 root root32097 May 13 06:14 my_sys.h
-rw-r--r--1 root root 2007 May 13 06:14 my_xml.h
-rw-r--r--1 root root 5841 May 13 06:14 raid.h
-rw-r--r--1 root root32081 May 13 06:14 readline.h
-rw-r--r--1 root root 3751 May 13 06:14 rlmbutil.h
-rw-r--r--1 root root 9687 May 13 06:14 rlprivate.h
-rw-r--r--1 root root 1385 May 13 06:14 rlshell.h
-rw-r--r--1 root root 2789 May 13 06:14 rltypedefs.h
-rw-r--r--1 root root 1751 May 13 06:14 sql_common.h
-rw-r--r--1 root root 6401 May 13 06:14 sql_state.h
-rw-r--r--1 root root 1055 May 13 06:14 sslopt-case.h
-rw-r--r--1 root root 1883 May 13 06:14 sslopt-longopts.h
-rw-r--r--1 root root  970 May 13 06:14 sslopt-vars.h
-rw-r--r--1 root root 3022 May 13 06:14 tilde.h
-rw-r--r--1 root root 1265 May 13 06:14 typelib.h
-rw-r--r--1 root root 1439 May 13 06:14 xmalloc.h

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can't build cdr_addon_mysql.

[Chris Mason (Lists)]

Marcel van Kaam, Fonetica wrote:


I had the same problem with installing addons. I checked out in the file
cdr_addons_mysql.c what the location of the asterisk.h must be and changed
the cdr_addons_mysql.c to the location of the asterisk.h file.

After this it worked. Also to be sure do: locate asterisk.h to check or you
have the file on your system.

Marcel 
 


Yes, that worked. For the record, it had to be

#include "../asterisk-1.0.9/asterisk.h"

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Visual ring notification

[Chris Mason (Lists)]

I have put a pbx into a resort with Polyycom phones, everythign works 
great, except the kitchen staff cannot hear the phone ring. I know many 
legacy systems employ a big red flashing light, any ideas on doing 
something similar?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Visual ring notification

[Chris Mason (Lists)]

I need one I can build into a kitchen hood and will be seen at 20'. 
Think fire alarm.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Visual ring notification

[Chris Mason (Lists)]

Just CYA regarding OSHA regulations on permissible noise levels.

 


OSHA who?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Provider Survey

[Chris Mason (Lists)]

List Receiver wrote:

Having used Broadvoice for a while with marginal service, I want to 
move on to another provider. So my question to the List is who is 
good? I know now one service is perfect but somebody out there has to 
be decent. Who have you guys had the best luck with?  

I suggest, if your installation is mission critial at all, you use a 
dialplan setup that has failover between two providers with failover to 
PSTN, and that you consider Teliax and NuFone as Providers. I have found 
these are serious players, they are not lemonade stands, and they offer 
failover to pstn for incoming calls also. I have never had an instance 
where both are down, but my experiene is relatively short.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Buy IP address

[Chris Mason (Lists)]

chawki hammoud wrote:


Hi:

I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or recommendations please?

 

I suspect what you are really asking for is a way to gateway a DID to 
your server. Having a dynamic IP is fine is you use a register statement.

You can use dynamic DNS to provide a stabel address.

Buy a DID from a provider that will accommodate a dynamic IP address via 
a register statement, most will.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Extensions will not go to voicemail

[Chris Mason (Lists)]

I have a remote installation that connects via IAX from my office pbx. 
When I call an extension on the remote pbx, after the dial period, the 
call is terminated. Nothing I do in configuration of that extension 
seems to matter:


   -- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-5", ""Dial 
710"") in new stack
   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-5", 
"SIP/710|30|tr") in new stack

   -- Called 710
   -- SIP/710-4841 is ringing
 == Spawn extension (office, 710, 2) exited non-zero on 
'IAX2/[EMAIL PROTECTED]:4569-5'

   -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5'

and the dialplan for that context is

[office]

exten => 710,1,NoOp("Dial 710")
exten => 710,2,Dial(SIP/710,30,tr)
exten => 710,3,Voicemail(u710)
exten => 710,103,Voicemail(b710)

Any ideas why I am not getting to the voicemail for that extension?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Long delay via Teliax

[Chris Mason (Lists)]

Joseph wrote:


I'm testing Teliax tall free number line and I'm experiencing long delay
about 1sec. during conversation.
When I call myself over FWD the response is normal no delay or cut
messages.
When I call my number over FWD the is a long delay, welcome message
usually cuts off few first words and during conversation my voice
arrives after about 1sec. delay.  
Since, the 800-number is only accessible from USA and I'm in Canada, the

only way I can test it is by calling it over FWD.

I've tested codec: ulaw and gsm
What might be causing such a problem?

 

I would analyse your network path for latency and packet loss, monitor 
it constantly and see if there is any correlation.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does Vonage support fax machines?

[Chris Mason (Lists)]

What makes you think they do? The marketing pieces? We all know G71 is 
not reliable for faxing, and for Vonage to advertise it is irresponsible 
of them.





 
Yahoo! Sports 
Rekindle the Rivalries. Sign up for Fantasy Football 
http://football.fantasysports.yahoo.com

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 




--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does Vonage support fax machines?

[Chris Mason (Lists)]

I've got a handful of ATAs  that support two ports... 

The question is, who make s areliable T.38 ATA? I mean one that can be 
expected to wrk with most gateways, in the way we have become used to 
G711/G729 ATAs working. Is there such a product?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Simpletelecom dead?

[Chris Mason (Lists)]

Bruce Ferrell wrote:


I've gotten word from their Marketing VP.  They are doing some kind of
massive move and expect to be down until Thursday



I smell lemonade...

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does Vonage support fax machines?

[Chris Mason (Lists)]

Before you give up, I have had good results with a Sipura 2002 ATA and 
using Teliax for faxing, I tried other termination accounts with the 
same setup and it didn't work.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Users handbook

[Chris Mason (Lists)]

Mark Phillips wrote:

This is somewhat unique to the site installation. For example, I don't 
have *69 programmed at my site because frankly there's no need for it 
with the Cisco 7960's.


I do however have an automatic conference booking utility and a 
speaking clock. Not often found in smaller sites.


I think you are on your own here.

If one is implementing an Asterisk solution in an office scenario, it 
has to have fairly similar features to another Asterisk installation. 
It's easy enough to edit and remove the parts that are different. What I 
am suggesting is a comprehensive "Here's everything Asterisk can do out 
of the box" document, change or remove what doesn't apply.


Let me know if any of you want to pool the work we have already done, I 
will compile to a complete document and post on the wiki.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom distributor in the UK ?

[Chris Mason (Lists)]

John Daragon wrote:


Hi;

I'm looking for a Polycom distributor in the UK who can supply a small 
number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ?


jd


I have been buying from Zycko - very efficient and on the ball.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's

[Chris Mason (Lists)]

Asterisk/phones work perfectly within our LAN.  Asterisk box has a public IP
- no NAT or firewalls.   When I take the phones to a remote location (again,
public IP - no NAT or firewalls that I know of) the outgoing audio does not
work.  I can hear the other party, my phones ring, I can dial out, etc, but
the other party cannot hear me (even if I dial #'s, etc).

Any ideas?

Thanks,
 

I had the same problem with a connection over our local Cable Co., even 
the engineers could not tell me why. I was able to route around it by 
putting in a direct route, anything that went through their gateway 
didn't work as described. Some internet routers and gateways drop rtp, I 
think, expecially systems designed to filter the traffic.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Any suggestions for an IP phone?

[Chris Mason (Lists)]

Alexandre Leclerc wrote:


Hi all,

We are in the process of selection IP Phones to work with our *new*
Asterisk PBX.

We want to buy 4 for something less than 1000$ but with a nice set of
features to work with our mail box, lines, good sound quality, full
duplex (and maybe speaker phone).

Any suggestions for something with good voice quality and not much
troubles to setup with Asterisk?

Voici quality is the most important point.

Thanks for any sugestion.

 

You'll love the Polycoms, the IP600 is amazing. Get one on your desk and 
you won't want to let it go.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Any suggestions for an IP phone?

[Chris Mason (Lists)]

You can go with the GXP-2000 from Grandstream.  It is shaping up as a
very good phone.  It usually retails for around $114 USD but you can
find it a bit cheaper if you really look around.

 

Two questions: Are the buttons rubbery, and is the display dim? I don't 
have the opportunity to test one and I have a bunch of Sipuras I don't 
need because of these failings.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Any suggestions for an IP phone?

[Chris Mason (Lists)]

[EMAIL PROTECTED] wrote:

Polycom does not support Asterisk. 
Thsi does not mean phones do not work with it.


Rudolf
P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones 
always say "can not contact boot server". Phones are set to use tftp and 
correct boot server IP is set via dhcp.
I will investigate further, but any suggestions are appreciated.

 

I always use FTP instead, it works famously. Make sure you configure the 
ftp server in DHCP or in the ftp servers settings, as an IP of course, 
and that you change the ftp password to the password for the user 
PlcmSpIp on the server.


After that it's flawless.

>Polycom does not support Asterisk.
Polycom, the company, does not support the use of the phones with 
Asterisk. Who cares? SIP is a standard, we don't need any help from them 
and we don't need their blessing. The phones are excellent quality and 
work very well with Asterisk, there's no support issue.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Mixed Voice/Data T1

[Chris Mason (Lists)]

We have a server running Asterisk and shorewall, three network 
interfaces and a T1 card, it functions as our firewall, pbx and connects 
to an Adtran 600 for FXS/FXO. We currently have two internet feeds, 
hence the three NICs.
Our 10 PSTN lines are currently delivered POTS to our FXO and rather 
lousy service, volume is low and voice quality is dull.


The Telco is offering me a Fractional T1 for our data needs. I thought 
that the best way would be to have the voice and data delivered on the 
T1. How difficult would it be to take the T1 into our Sangoma T1 card 
and seperate out the voice to asterisk and the data as our internet feed?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Polycom behind firewall issue

[Chris Mason (Lists)]

I have a user that just got a broadband connection so she could have an 
extension off our pbx. The service is DSL and uses a speedstream 5200 
dsl router. I sent her a Polycom IP300. At first it would not access the 
config files via ftp so I had tech support walk her through setting the 
phone's internal IP to be the dmz. This allowed me to set up the phone 
using the web interface and now it registers. We had NAT problems so I 
set the NAT features of the phone:

IP Address: 67.136.nnn.nnn
Signalling Port : 5060
Media Port Start: 1

In sip.conf, I have
nat=yes
externalip=67.136.nnn.nnn
qualify=yes

I can call the user and she can hear me. If she calls me, no voice can 
be heard either way. When I run sip show channels, I see:


Peer User/ANRCall ID  Seq (Tx/Rx)   Format  Last Msg
192.168.0.169805 02fe2b2a684  00102/0   g729Tx: ACK
67.136.nnn.nnn893 8dae34ea-ae  00101/1   g729Rx: INVITE
67.136.nnn.nnn(None)  3926de51-a1  00101/1   unknow  Rx: 
REGISTER



and it just stays like that until the call is terminated. I would think 
it was an rtp / nat problem, any ideas how to fix?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom configs?

[Chris Mason (Lists)]

Michael Graves wrote:


I have a number of Polycom phones to setup with my * server. For my
initial few phones I hand wrote configs. Does anyone here who uses
Polycom phones have some form of management utility for automating
their setup?

 

I wrote myself a very simple script that makes provisioning the phone a 
one line command. Let me know if you would like it.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Voicemail macro?

[Chris Mason (Lists)]

For our hotel application, we don't want to have to write 50 voicemail 
entries, is there a way to do a voicemail macro in the same way as a 
standard extension macro?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Voicemail management

[Chris Mason (Lists)]

For our hospitality system, voicemail management is an issue. I looked 
at vmail.cgi and it works for the user, but I need a higher level 
management capabikity, i.e., flush all email from extensions 1XX 
(Apartment1) when a guest checks out.

Is there anything like that or does anyone want to work on this for me?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Any Ideas??? 3rd time posting => Sipura SIP Phones Multi-Line

[Chris Mason (Lists)]

Steve Gladden wrote:


Still looking for some direction with this subject:

I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it

This is where you have several sipura-841 SIP phones for example... if
someone pickes up 'line1' I'd like the light to come on on ALL
phones to indicate someone is using 'line 1' and they should NOT be able
to pick up 'line 1' so long as that 'line1' is in use by another phone.
 

You are trying to emulate a key system, Asterisk is a PBX. I don't you 
will get this done as there is no concept of lines, all phones are 
extensions.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] HOW TO make xten eyebeam incoming video start before you send yours

[Chris Mason (Lists)]

Matt Riddell wrote:

Sorry, I don't remember who was asking about this, but it seems that 
if you record a video message that contains the send video start, it 
will actually fire up the remote receive window.


I.E. Previously I was using the recording section of voicemail to 
create my video IVR's.  This meant that when I arrived at the section 
to record the message, I had already clicked send video in Xten.


This meant that when you dialed an extension which played back these 
files, the video wouldn't start unless you sent video first.


Now, if you don't click send video in Xten until after you hear the 
beep for recording, it works.  It will play you the video just by 
dialing the extension.


I'm using the latest beta of eyebeam by the way.


Matt,
Where did you get teh beta from? I purchased the release version but I 
would like to have the same version as you to get it working wth Asterisk.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Best VoIP provider

[Chris Mason (Lists)]

Madhawa Jayanath wrote:


o Bernie,
1) best results www.nufone.net
2) low cost www.voipjet.com


Anyone able to find NuFone's rates? I have been looking for them on 
their site. I need international rates and UK Mobile.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Polycom phone configuration script available for download

[Chris Mason (Lists)]

I have tidied up the script and added some help text, feel free to 
download and maybe improve.

http://www.masonc.com/phoneconf

Usage: Usage: ./phoneconf [config|help] phonemacaddress extention 
username context


./phoneconf help will print syntax info
./phoneconf 0004f201aa11 500 MyPhone defaulot
will configure the  phone with that mac with one extension (500), and 
add the phone to sip.conf.
Edit at least the first two variables before using and change the 
sip.conf stuff to suit, this works well for me on internal and remote 
phones.

Improvements greatly appreciated, I'm a crappy programmer.
** Use at your own risk **

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Interconnect with Mitel PBX

[Chris Mason (Lists)]

I have a small government department that wants me to implement a 
Asterisk installation, however, they connect to the Government PBX, a 
Mitel SX200, and want to keep the ability to do that. I know there is no 
chance to connect the digital extension lines, but would it be possible 
to have the pbx admins send analogue extensions over and have those 
lines interface through an FXO interface? Or what other way could it work?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: Polycom 600 one-touch message access?

[Chris Mason (Lists)]

Yes, I have that setup too (no change from 1.4.1)

Are you saying one-touch voicemail works for you with 1.5.2 ?

(meaning no message count summary screen when pressing "Messages")

 

If you have more than one line, you will get a summary, otherwise, how 
would you access the other lines voicemail?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Mixed Voice/Data T1

[Chris Mason (Lists)]

Adam Dobrin wrote:

As nice as HDLC sounds in theory; we have the same setup, a T1 with 
afew lines split off, and i just don't see a need to add the routing 
load to the asterisk machine.  We have an Adtran 604 which splits the 
T into a PRI and 10/100.  Incidentally, HDLC in asterisk seems to be.. 
a hassle to get working at best.



Sangoma tell me that their T1/E1 cards  are specifically designed to 
this and it will work well. My telco doesn't have the ability to deliver 
the data and voice over a T1 so it's moot, we will not be going this 
route, but it looked like a perfect solution.



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users