Re: [asterisk-users] SPA942 quality for a Bank

2006-10-15 Thread VoIP Street

Rich Adamson wrote:
Before committing to about 50 of the spa942's, I like to take a last 
poll from those on the list to identify any negative issues that might 
be associated with the audio, functionality, early failures, etc, on the 
spa942.


Expecting to deploy these using existing cat5 cabling and both rj45 
jacks. Been using three of theme in a short term demo with the customer, 
but the demo systems has been purposefully configured with only basic 
telephony functions.


Oh... someone mentioned the headset (no handset) pin jack is only for 
the microphone (and not the speaker) which would seem very odd. Anyone 
using a headset with the 942 where both the microphone and earpiece 
function fully?


Any thoughts?

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We use SPA 941's internally and are very happy with them.. Headset jack 
works great. Speaker premo.. Could use a backlight but other than that 
we are very happy.. You couldn't pry mine out of my cold dead hands..



--
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Re: [asterisk-users] DID failover

2006-10-15 Thread VoIP Street

Todd- Asterisk wrote:
I'm setting up an asterisk server where an administrator will not always 
be available in case of problems.  


**SNIP**

Not sure about your provider but within our control panel we offer our 
customers the ability to set multiple devices. Devices are basically 
setup as a one to one relationship to your Asterisk servers and or IP 
phones, soft phones etc.


So you could setup 2 devices in our control panel or with your provider 
if they offer this.


Device 1: Asterisk Server (Primary)
Device 2: Asterisk Server (Secondary)

Each device creates a SIP or IAX username and password and ability for 
each machine to register to us.


When it comes to DIDs you can configure them in the control panel to all 
point to Device 1, in the event of a failure on device one you simply 
log in to our control panel and repoint your DIDs to Device 2. It takes 
approximately 5 minutes for our queue to run and all calls will now flow 
from your DIDs to Device 2. At any time you can push calls out through 
Device to and terminate to us regardless of when/if you repoint your DIDs.


In the case of getting your internal phones to re-route to the correct 
system.. Maybe you could leave server 2 off, pre-configured with the IP 
of server 1. If you have a failure, simply turn off server 1 at IP 
X.X.X.X if it's not already dead and have someone hit the power switch 
to Server 2 which will now be at the same X.X.X.X IP. No IP phone 
re-configuration required.


I am sure there is a more elegant solution using a router or something 
but, this is just one approach.


So, there are a few ways to skin that cat, just up to you how you want 
to go about it.


Hope this helps..


--
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Re: [asterisk-users] Setting CALLERID on a residential telco line

2006-08-04 Thread VoIP Street

hugolivude wrote:

That's what I feared.  I could do it if I had a T1 is that right?

Thanks,
H

On 8/4/06, Steven Ringwald [EMAIL PROTECTED] wrote:

hugolivude wrote:
 Redhat 9
 Asterisk - 1.2.7
 TDM 400 - 1 FXO, 2 FXS

 I'm using a standard residential PSTN line on my ZAP channel and
 curious whether I can override the caller ID my telco has for me with
 one of my choosing.

 I've tried this:

 exten = s-ZAP,n,Set(CALLERID(all)=My Name 999-999-999)
 exten = s-ZAP,n,Dial(Zap/g2/6137451576)

 but the callee still sees my telco callerid.  Have I missed something
 or does the telco ultimately control CallerID on a residential line?
 It stands to reason it would, but I'm hopeful I'm wrong!!


If it is a POTS line, you cannot change the caller*id.

Steve

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That's correct, as long as your provider allows it. You can also do it 
with SIP and IAX providers most of the time.


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Re: [asterisk-users] phpagi problem

2006-07-18 Thread VoIP Street

Mauricio Mantilla wrote:
Yeah, I wrote it on a different file and it worked, i guess there was 
some hidden character in the script that was messing it up.

Thank you anyway,

Mauricio Mantilla

On 7/17/06, *Time Bandit* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


  #!/usr/bin/php -q
   ?php
  require('/var/lib/asterisk/agi-bin/phpagi.php');
  $agi = new AGI();
  $agi-say_digits(62410);
  $cid = $agi-get_variable(dir);
  $agi-say_digits($cid);
   ?
 
 
  I'm getting this error:
 
  parse error, unexpected '=' on line 6

I don't know why you're getting this error, it parse correctly here.

But one thing is that the line $agi-say_digits($cid); won't work.
When you do a get_variable, the result you get is an array, and the
member holding the value is 'data'. So you have to write your line
like this : $agi-say_digits($cid['data']);

Check the documentation : http://phpagi.sourceforge.net/phpagi2/docs/

hth
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dos2unix usually fixes these sorts of issues.

Just do something like this at the command line:

dos2unix filename filename

It removes a lot of the garbage that causes trouble with command line 
PHP scripts and AGIs etc.



--
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Re: [asterisk-users] How many users on an asterisk box behind a dsl can you have

2006-07-17 Thread VoIP Street

ted jones wrote:
I have been trying to read up and understand Asterisk.  I have a small 
office of 25 people growing to 50 and have a dedicated DSL for Asterisk 
and another DSL for computer use and was wondering using gsm primarily 
how many users I could put on the asterisk box on a single dsl.  Average 
calls is probably going to be 25-35 at any given time.  Any help or 
suggestions would be appreciated.
 
Ted
 



See the all-new, redesigned Yahoo.com. Check it out. 
http://us.rd.yahoo.com/evt=40762/*http://www.yahoo.com/preview





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Much would depend on the specs of your DSL line and the hardware you 
plan to run your * server on.



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Re: [asterisk-users] Problem with making outgoing calls

2006-07-12 Thread VoIP Street




Crazy Boy wrote:

  Hi,
  
  We could
makecalls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have
realized that Teliax server was down for few hours. 
  
  Currently
our Asterisk server is connecting with Teliax. But, When I am trying to
make call to USA, Its giving me one ring and being disconnected. I
could not understand what could be the problem? Isthere any problem
with my connection to Teliax server?
  
  
  With the
configuration mentioned below, we could successfully make calls to USA.
But, from this morning we are unable to make calls to USA. 
  
  IAX.CONF
file contents:
  
  disallow=all
  allow=ulaw
  
  SIP.CONF file contents:
  
  [101]
  
  type=friend
  username=101
  secret=abcd
  callerid="Ani"
  host=dynamic
  context=tutorial
  
  [general]
register = ab.cd:[EMAIL PROTECTED]
 
[authentication]
auth = ab.cd:[EMAIL PROTECTED]
  
[teliax]
context=default
  type=friend
username=ab.cd
user=ab.cd
host=voip-co1.teliax.com
secret=xx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
  
  
  EXTENSIONS.CONF
file contents:
  
  exten
= 101,1,Dial(SIP/101,15)
  
  exten
= 101,2,Voicemail(u101)
  
  exten
= 101,3,Voicemail(b101)
  
  exten
= 101,4,Hangup
  
  exten =
_1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
  
exten = 3031234567,1,Answer()
exten = 3031234567,2,DIAL(SIP/user,20)
  
  
  VOICEMAIL.CONF file
contents:
  
  101 = ani, Ani, [EMAIL PROTECTED], [EMAIL PROTECTED]
  
  
  
  
  Please
let me know the problem ASAP. Looking forward to your response. 
  
  Thank you.
  
Regards,
Chandra.
  
   
  Do you Yahoo!?
Get on board. You're
invited to try the new Yahoo! Mail Beta.
  

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Do you get some sort of error if you look at the console (CLI) while
attempting the call? If not, give it a try.. If you try and don't see
anything useful, try turning on debug by typing "sip debug" if using
sip or "iax2 debug" if using IAX.

When you are done use "sip no debug" or "iax2 no debug" to turn it off
:)

--
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Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com



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Re: [asterisk-users] Email notification of voicemail

2006-07-12 Thread VoIP Street

Kevin Savoy wrote:
Asterisk is trying to send an email to users when they receive a 
voicemail. Can this be shut off? I have not entered any email addresses 
in voicemail.conf so it tries to send to [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]. This of course gets rejected 
since the user does not exist and the root users mailbox on linux gets 
full of these rejection notices. I can’t seem to find anywhere to tell 
Asterisk to stop notifying people they have voicemails.


 


I’m using 1.2.9.1 of Asterisk. Thanks

 


_

 


**Kevin Savoy**

**Business Unit Telecom Analyst**

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 





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You could try commenting out:

attach=yes

Also, if you don't want any emails sent ever for any voice mail users 
you could probably uncomment the following line and give it a bogus path 
to the mailer.


;mailcmd=/usr/sbin/sendmail -t

There is probably a better way to do this but we have never needed to 
turn it off so I am not sure.


Hope this helps.

--
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Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com
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Re: [asterisk-users] Email notification of voicemail

2006-07-12 Thread VoIP Street

Kevin Savoy wrote:

I have attach=no in my voicemail.conf so that can't be doing it. Not sure
where that sendmail command is. Don't see it in voicemail.conf or any other
config in the asterisk directory.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street
Sent: Wednesday, July 12, 2006 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail

Kevin Savoy wrote:
Asterisk is trying to send an email to users when they receive a 
voicemail. Can this be shut off? I have not entered any email addresses 
in voicemail.conf so it tries to send to [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]. This of course gets rejected 
since the user does not exist and the root users mailbox on linux gets 
full of these rejection notices. I can't seem to find anywhere to tell 
Asterisk to stop notifying people they have voicemails.


 


I'm using 1.2.9.1 of Asterisk. Thanks

 


_

 


**Kevin Savoy**

**Business Unit Telecom Analyst**

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 





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You could try commenting out:

attach=yes

Also, if you don't want any emails sent ever for any voice mail users 
you could probably uncomment the following line and give it a bogus path 
to the mailer.


;mailcmd=/usr/sbin/sendmail -t

There is probably a better way to do this but we have never needed to 
turn it off so I am not sure.


Hope this helps.

I got this from the voicemail.conf file on a server running 
CVS-NHEAD-04/13/05-00:37:14


Not sure if it's still valid in later versions.

--
VoIP Street
Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com
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Re: [Asterisk-Users] Supporter needed

2006-06-07 Thread VoIP Street .com

Soren,

We do not do this kind of work but know a consultant that may be 
interested. We can recommend him highly as he has worked with us in the past 
on some projects.


I will forward your information to him and ask him to contact you.

--
VoIP Street
DID origination services
with support you can count on!
http://www.VoIPstreet.com

- Original Message - 
From: Soren Christensen [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, June 07, 2006 12:22 PM
Subject: [Asterisk-Users] Supporter needed



Hi,

I'm looking for a great tech support person to take over the admin of our 
asterisk system. If you are a networking person as well, with some 
experience in firewalls and desktop support even better. The system is a 
multi-group system with IVR, Follow-me dialing, voicemail, and 
conferencing. Multiple SIP providers are in use.


If you feel you can help us, or can recommend someone that can - please 
let me know. Thanks


We are located in Menlo Park, CA.

/S


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Re: [Asterisk-Users] Asterisk Radius Module

2006-05-28 Thread VoIP Street .com



How about this one?
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth

--VoIP StreetDID origination serviceswith support you can count 
on!http://www.VoIPstreet.com

  - Original Message - 
  From: 
  Oliver 
  Vermeulen 
  To: asterisk-users@lists.digium.com 
  
  Cc: 'Commercial and Business-Oriented 
  Asterisk Discussion' 
  Sent: Sunday, May 28, 2006 4:09 PM
  Subject: [Asterisk-Users] Asterisk Radius 
  Module
  
  Hi 
  List,
  
  I'm looking for a 
  Asterisk radius module ... Anybody has one ?
  
  Thanks,
  Oliver
  
  
  

  

  
  Oliver 
  VermeulenWorld Venture Group 
  Telecom 
  
  Corporate 
  Address:Str 
  Avionului 
  Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: 
  +(40)31-860-0030Fax:  +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK 
  DID:+(44)870-478-8896SIP 
  : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com
  
  
  

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Re: [Asterisk-Users] Asterisk Radius Module

2006-05-28 Thread VoIP Street .com



Never used it, but I knew some sort of Asterisk 
radius thing existed so I searched the Wiki for it and replied. Sorry it doesn't 
work for you, good luck with your search.

--VoIP StreetDID origination serviceswith support you can count 
on!http://www.VoIPstreet.com

  - Original Message - 
  From: 
  Oliver 
  Vermeulen 
  To: 'VoIP Street .com' 
  Sent: Sunday, May 28, 2006 9:13 PM
  Subject: RE: [Asterisk-Users] Asterisk 
  Radius Module
  
  I need full AAA
  
  That just dose authentication..
  
  But i will try it lets see ;)
  
  Are you using it ?
  
  Oliver
  
  
  From: VoIP Street .com 
  [mailto:[EMAIL PROTECTED] Sent: Monday, May 29, 2006 3:59 
  AMTo: [EMAIL PROTECTED]; 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Asterisk Radius Module
  
  How about this one?
  http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
  
  --VoIP StreetDID origination serviceswith support you can 
  count on!http://www.VoIPstreet.com
  
- Original Message - 
From: 
Oliver 
Vermeulen 
To: asterisk-users@lists.digium.com 

Cc: 'Commercial and 
Business-Oriented Asterisk Discussion' 
Sent: Sunday, May 28, 2006 4:09 
PM
Subject: [Asterisk-Users] Asterisk 
Radius Module

Hi 
List,

I'm looking for 
a Asterisk radius module ... Anybody has one ?

Thanks,
Oliver







Oliver 
VermeulenWorld Venture 
Group Telecom 
Corporate 
Address:Str 
Avionului 
Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: 
+(40)31-860-0030Fax:  +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK 
DID:+(44)870-478-8896SIP : 
[EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com




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Re: [Asterisk-Users] AstriCon

2006-05-26 Thread VoIP Street .com
We attended the Astricon in California, US last year. Although it was not 
what we expected, we did feel like we gained enough knowledge to make it 
worth the time and expense to attend.


Good luck and let us know how you like the show if you end up attending.

--Todd


--
VoIP Street
DID origination services
with support you can count on!
http://www.VoIPstreet.com

- Original Message - 
From: Mimmus [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Friday, May 26, 2006 9:06 AM
Subject: [Asterisk-Users] AstriCon



Hi,
I live in Italy and I'm planning to go to the next AstriCon conference in
London.
Can someone on this list provide me with some detail of previous 
exhibition?

I'd like to have some idea of what I'm running into...

Thanks
--
Domenico Viggiani

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