RE: [Asterisk-Users] open source sip softphone (Window OS version )

[Kerry Garrison]

None of those are open source that I recall.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Derek Whitten
> Sent: Thursday, June 15, 2006 6:07 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] open source sip softphone 
> (Window OS version )
> 
> Asterisk guy wrote:
> > are there any open source sip softphone (Window OS version )?
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> 
> xlite
> sjphone
> firefly (3rd party version)
> 
> 
> 


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[Asterisk-Users] Voicemail to Email on Blackberry

[Kerry Garrison]

Is there any setting 
in the voicemail that will send the voicemail file in a type that is recognized 
on a Blackberry?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] GXP-2000

[Kerry Garrison]

With hundreds of installed phones now, here are my choices in order

Linksys SPA-941/942
Polycom 501/601
Cisco 7960
Polycom 301
Snom 320/360

I would never ever ever sell a client on a SPA-841 or heaven forbid the
GXP-2000. All the clients who bought those originally sold them off and went
for better phones very quickly.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com




> Polycom 501
> Linksys spa-941
> Polycom 301
> Sipura/Linksys spa-841
> Grandstream GXP-2000
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RE: [Asterisk-Users] TDM2400P with echo canceller not working

[Kerry Garrison]

If you call Digium they will help you get the card configured properly. You
get installation support with any of their hardware products. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Giorgio Incantalupo
> Sent: Monday, May 29, 2006 12:33 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] TDM2400P with echo canceller not working
> 
> Hi,
> I have  a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 
> 1.2.1) and a TDM2400P with echo canceller. I installed the 
> card but no echo cancellation is being made...seems like the 
> echo canceller module does not work, infact the software 
> cancellation is working.
> 
> My zapata.conf has echocancel = 128 and echocancelwhenbridged 
> = yes but no echotraining parameter which gives a warning.
> 
> I found no info about how to use this card and how to 
> correctly set zapata.conf, which zaptel version to use, etc...
> 
> Does anybody knows how to use this card?
> 
> TIA
> 
> Giorgio Incantalupo
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RE: [Asterisk-Users] features

[Kerry Garrison]

To enable call waitng on 
extension 105
 
database put CW 105 
ENABLE
to disable call walting 
on extensions 105
 
database put CW 105 
DISABLE

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Khaled 
  ChehabSent: Saturday, May 27, 2006 1:43 PMTo: 
  asterisk-users@lists.digium.comSubject: Fw: [Asterisk-Users] 
  features 
  
  
  

 Dear 

 
In need to know how 
to add an asterisk feature as call waiting (*70) or disable call waiting 
(*71) for a user with out using a sip phone (DTMF)or gateway I want to 
manage it form a command line  or a Perl script or php web interface 

 
Or editing a file 
at asterisk directory .
 
 
In need to it 
urgently 
 
 Regards
Thanks 


  
   
   
  
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  authorized to conclude any binding agreement on behalf of Xplorium with 
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RE: [Asterisk-Users] Web based interface

[Kerry Garrison]

There are several listed at http://voip-info.org. For Management check out
FreePBX, for recorded calls look for Asterisk Recording Interface.
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Kanishka Somaratne
> Sent: Saturday, May 27, 2006 9:55 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Web based interface 
> 
> hello
> is there a web based interface for IVR management, check 
> voice mail, check recorded calls and ect.
> 
> regards
> kani 
> 
> 
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RE: [Asterisk-Users] FreePBX virtualization

[Kerry Garrison]

You can by creating different contexts and using the Administrators function
allow them to modify some of the settings themselves.
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Daniel Salama
> Sent: Thursday, May 25, 2006 10:42 AM
> To: Non-Commercial Discussion Asterisk
> Subject: [Asterisk-Users] FreePBX virtualization
> 
> Does FreePBX support virtualization of its services? For 
> example, can I use it to provide virtual PBX to different 
> clients under the same instance of FreePBX? Or is it more 
> geared to single office-type installation?
> 
> Thanks,
> Daniel
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RE: [Asterisk-Users] Is NuFone Really Dead? Same as voipjet

[Kerry Garrison]

This is the same with VoipJet, some people have good luck but my lines have
been down for 3 months and all attempts at contacting them have gone
unanswered. Hard to believe people still rave about their service. 

Here is a hint folks, if the company does not post a customer service PHONE
NUMBER don't use them. Secondly, if they do have a phone number but nobody
ever answers it, don't use them.

Just because their email address is [EMAIL PROTECTED] doesn’t mean its
fast, or is even answered. It should be /dev/[EMAIL PROTECTED]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Carlos Chavez
> Sent: Thursday, May 25, 2006 7:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Is NuFone Really Dead?
> 
> On Wed, 2006-05-24 at 14:00 -0700, Andy Jefferson wrote:
> > Went to their site today. Site claims they are still in 
> biz. What is 
> > the story? What really happened to Nufone anyway?
> 
>   They say they are but I have 2 800 DIDs with them that 
> are still offline, they stopped working more than a month ago 
> and all attempts to contact support have been unanswered. I 
> guess it is up to luck, some users have service others do 
> not.  Do you really want to work with a company like that?
> 
> --
> Carlos Chavez Prats
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
> 


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RE: [Asterisk-Users] What and When is the next version of Asterisk?

[Kerry Garrison]

1.2.8 would be the logical next version.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Obelix
> Sent: Wednesday, May 24, 2006 1:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] What and When is the next version 
> of Asterisk?
> 
> 
> What and When is the next version of Asterisk?
> 
> /Obelix
> 
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RE: [Asterisk-Users] latest @Home questions

[Kerry Garrison]

You will have far better luck asking this in the AAH forum or the FreePBX
site.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael George
> Sent: Wednesday, May 24, 2006 1:42 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] latest @Home questions
> 
> We are moving our asterisk 1.0 system to a new Asterisk @Home 
> system (2.8) and I am the one in charge of doing it.
> 
> I have run into a snag, though, on meetme conferences and 
> with the transfer key.
> 
> Regarding the transfer, it appears that both directions of 
> all calls can transfer by pressing the # key.  I do not like 
> that ability.  I would like to change it by doing 2 things:
> 
> 1. Make the transfer sequence be ## rather than #
>   I looked at the features.conf file and it didn't have 
> an entry for
>   blindxfer, so I added it.  However, # is still the 
> transfer character
>   so it doesn't seem to be recognizing the settings from
>   features.conf.
> 
> 2. Not allow incoming calls to transfer at all.
>   I've looked at the dial() string on incoming calls and 
> they do not
>   contain a t or T like I would expect for the channel to 
> be able to
>   transfer.
> 
> As for the meetme conferences, the docs say that for all 
> extensions defined, there is a meetme conference at 8.  
> So extension 250 would have a conference at 8250.  This isn't 
> the case on our installation.
> 
> I went to the Conference menu item and defined a conference 
> at 1000 and I put an entry into the IVR for incoming callers 
> to get into the conference and that works fine (except that 
> after the PIN number, they cannot press # to signal the end 
> of the PIN -- that will try a transfer).  However, I don't 
> have a way to get to the conference from an extension.
> 
> I think I'm missing something, because meetme setup cannot be 
> that difficult...
> 
> Thanks for any help anyone can offer.
> 
> --
> -M
> 
> There are 10 kinds of people in this world:
>   Those who can count in binary and those who cannot.
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RE: [Asterisk-Users] What about T400 T1 cards?

[Kerry Garrison]

Little personal preferance here (and hopefully some payback for some help
you gave me a while back). My experience has been that unless you have echo
cancelation on the hardware, the hardware isnt worth purchasing.  This holds
for the TDM and T cards right from Digium. If I am going to do a T1 install,
I insist on the high end quad port card because it is the only one with
hardware echo cancelation. Other good cards to look at are the Sangoma and
the Rhino of which I have had good results from as well. But the low-end
Digium and so far, any clone card, aren't worth the PCB they are printed on.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Bart Fisher
> Sent: Tuesday, May 23, 2006 7:49 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] What about T400 T1 cards?
> 
> Can anyone clue me in about these T400 T1 cards I see 
> advertised?  I hear they are Digium Clones.  Is there some 
> reason to avoid these?  How do they compare to TE410P's for example.
> 
> Bart 
> 
> 
> 
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RE: [Asterisk-Users] Best VoIP provider for Asterisk

[Kerry Garrison]

Depends on your location and your requirements. A generic 
post like this generally turns into a flame war. Please be MUCH more 
specific.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
  BoySent: Tuesday, May 23, 2006 5:56 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Best VoIP 
  provider for Asterisk
  Hi Friends,Can you please tell me who is the best VoIP 
  Service Provider using Asterisk (With trail version for sometime) . Waiting 
  for your quick response. Thank you.Regards,Chandra.
  __Do You 
  Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around 
  http://mail.yahoo.com 
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RE: [Asterisk-Users] US telco lingo

[Kerry Garrison]

 

Could someone explain to a non-US dummy the following phrases I have
seen on the list.

"I can provide you with tier 1 termination 6/6.  I can blend or
NPANXX breakout."

"We provide US48 termination, blended rate for 1 MOU and above is
.008 with 6/6."


What is 6/6?
2/3 devil?
Normally I would take that to be minimum 6 second billiing and
billed in 6 seconds increments.
 
What is US48?
Contentinal US, lower 48, all states by Alaska and Hawaii.

What is blended?
What you do with ice, alchohol, and a mixer



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[Asterisk-Users] Watchguard Firebox 1000 woes

[Kerry Garrison]

We are trying to 
setup a sip connection behind a Watchguard Firebox 1000 and it is simply 
put...not working. The ports are all forwarded but the packets are not going 
out. It is as if the firewall simply ignores SIP packets. Has anyone seen this 
or have any idea what the issue could be? Watchguard so far has been of zero 
help.
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: Asttapi for Asterisk 1.2 Testers Needed (was RE: [Asterisk-Users]Asterisk TAPI - Outlook click2dial)

[Kerry Garrison]

Please hook me up. I have customers dieing for something that works. All our
systems are Asterisk 1.2.7.1
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Clint Sharp
> Sent: Tuesday, May 16, 2006 11:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Asttapi for Asterisk 1.2 Testers Needed (was RE: 
> [Asterisk-Users]Asterisk TAPI - Outlook click2dial)
> 
> I've finished a patched version of asttapi that will work 
> with asterisk 1.2.  There were fundamental changes to the 
> Asterisk Management interface between 1.0 and 1.2 that broke 
> asttapi.  I think my patched version will work on 1.0 and 1.2 
> branches, but I have no way of testing since I don't have a 
> 1.0 install nor do I want one :).
> 
> I'm looking for testers, if anyone's willing to test this 
> out, I'll send you a zipped copy of the TSP file (I haven't 
> worked on doing an installer yet).  I need to send out the 
> debug build so I can generate information in case it doesn't 
> work on anyone else's PC.  Contact me off-list for copies.
> 
> Clint
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tomislav Vojvodic
> Sent: Friday, May 12, 2006 1:39 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
> 
> Oh.. :/ too bad.. 
> 
> I'll have to look at the source.. 
> 
> bye,
> 
> 
> Tomislav
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw
> Sent: Thursday, May 11, 2006 11:20 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
> 
>  Yes, I have the exact same problem.
> :(
> 
> 
> -Original Message-
> From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 11, 2006 5:48 AM
> To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
> 
> Hey, thanks for your reply.. ;)
> 
> I'm also using asttapi from website you posted -> omniis.com. 
> 
> Version is 0.10 (newest)
> 
> Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup'
> isn't
> even implemented in AstTAPI driver so that could be the reason why
> Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. 
> 
> When I clik 'Hangup' in Outlook there is nothing in Asterisk 
> debug/cli window.
> 
> Only problem is that Outlook still thinks that call is active 
> even if you hangup the phone manually.. I mean, when I put 
> the earphone back to base/station/phone.. whatever. Dialing 
> works just fine.
> 
> Because of that you need to close that window 2 or 3 times if 
> you want to call same person/contact again.
> 
> Bye,
> 
> Tomislav
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of T.S
> Sent: Thursday, May 11, 2006 1:08 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
> 
> I had similar problems when I first started to play with it. 
> I've gotten Omniis TSP for Astrisk to work just fine. 
> http://www.omniis.com/asttapi But i don't know the version im 
> using 0.0.8
> 
> Terrelle
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tomislav Vojvodic
> Sent: Wednesday, May 10, 2006 2:23 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
> 
> Hello,
> 
> I'm experiencing some problems with AstTAPI driver. Dialing 
> works just fine, but 'Hangup' from Outlook doesen't.. 
> actually that's not the problem as fact that Outlook doesen't 
> detect end of conversation -> once the call is terminated 
> 'manually' via the phone Outlook still 'thinks' that call is active.
> 
> Anyone knows what's the problem? Is 'hangup' implemented in 
> AstTAPI driver?
> 
> Thanks,
> 
> Tomislav
> 
> 
> 
> 
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RE: [Asterisk-Users] Asterisk with SIPconnect

[Kerry Garrison]

We are a Cbeyond partner and have implemented their SIPConnect product. My
main complaint is that they don't let me spoof the outbound caller id yet.
They lock it down to one specific number. So if users with their own DID's
want their number to go out for caller id, you cannot do that at this time.
Otherwise it works great and we will continue to use it for future clients.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Brian Gorby
> Sent: Monday, May 15, 2006 12:17 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Asterisk with SIPconnect
> 
> Has anyone had any experience connecting Asterisk to 
> Cbeyond's SIPconnect service (http://www.sipconnect.info)? 
> Any opinions?
> 
> Thanks,
> 
> -Brian
> 
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RE: [Asterisk-Users] Voicemail WAV to PDA Problems

[Kerry Garrison]

Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without any
problem. My problem was that the MDA is a worthless turd and a complete joke
as a phone. I took it back and switched to the backberry 8700g which has its
own attachment problems. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Peder @ NetworkOblivion
> Sent: Friday, May 12, 2006 9:02 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Voicemail WAV to PDA Problems
> 
> Our asterisk server has been up and running for over a year 
> and it works great.  I have emails going to my account as an 
> attachment and I can listen to them on the desktop and it 
> works fine.  I just got a T-Mobile MDA that runs Windows 
> Pocket (or whatever they call it) and it can check email.  If 
> I have it download the email, it gets the attachment, but it 
> can't seem to play it (it CAN play wav files).  If I take the 
> email that was sent to my home account and then "forward it 
> to myself" and let the MDA pick it up, then it can play the 
> attachment.  So clearly it isn't an issue playing WAV's, or 
> even WAV's from Asterisk, it's some email attachment issue 
> with the way Asterisk or Postfix sends the attachment. 
>   Has anybody else run into this problem?  If so, any help 
> would be appreciated.
> 
> Peder
> 
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[Asterisk-Users] Linksys IP Device Bulk Provisioning Guide

[Kerry Garrison]

I have written up an 
guide on how to do bulk provisioning of the Linksys phones and 
ATAs.
 
http://voipspeak.net/index.php?option=com_content&task=view&id=73
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] How do I monitor the whole conversation on aZap channel ...

[Kerry Garrison]

How are you trying to do it? ChanSpy or ZapBarge? 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anthony Azzopardi
> Sent: Wednesday, May 10, 2006 9:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How do I monitor the whole 
> conversation on aZap channel ...
> 
> I want to monitor the zap channel while the phone is answered 
> by a human not by asterisk.
> Tzafrir Cohen wrote:
> 
> >On Wed, May 10, 2006 at 08:53:36AM +0200, Anthony Azzopardi wrote:
> >  
> >
> >>How do I monitor the whole conversation on a Zap channel without 
> >>answering it - the channel is hanging up, I think it's because it's 
> >>not answered.
> >>
> >>
> >
> >If the channel is not answered, there is no (useful) audio in it to 
> >monitor. What exactly are you trying to do?
> >
> >-- Tzafrir
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> >
> >
> >  
> >
> 
> 
> --
> Best regards,
> 
> Anthony Azzopardi.
> 
> Tel:  79713618
> Email:[EMAIL PROTECTED]
> 
> 
> Sign up for free voip from http://line.sytes.net
> 
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RE: [Asterisk-Users] VOIP provider

[Kerry Garrison]

Have you looked at CBeyond? I like their T1 SIPConnect 
product.

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Nitin 
  GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] VOIP provider
  
  Hi,
   I am looking for voip providers in bay area, any suggestions?
  My requirements are:
   handling around 2000 calls a day (incoming) and around 1000 calls a 
  day outgoing. I have a Asterisk PBX server to take care of routing calls to 
  appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support 
  from VOIP provider. Also a dedicated t1 line in case provider can provide this 
  too. 
   
  Thanks,
  Nitin
   
   
   
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RE: [Asterisk-Users] ethernet interface shares interrupts with tdm card

[Kerry Garrison]

Go into the BIOS and disable every possible device like 
USB, COM, Serial, etc. But odds are, you are screwed with that 
motherboard.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Antonio 
  AlmodóvarSent: Wednesday, May 10, 2006 8:01 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] ethernet interface shares interrupts with tdm 
  card
  Hello.I have a MinITX motherboard with only one pci slot and 
  one onboard ethernet interface, I have a TDM04B card plugged into that 
  motherboard and the 
  /proc/interrupts:CPU0     
  0:  169626332  
  XT-PIC  timer   
  1:   
  1270  XT-PIC  
  i8042  
  2:    
  0  XT-PIC  
  cascade  
  8:    
  4  XT-PIC  
  rtc 12:  
  170166219  XT-PIC  
  eth0, wctdm 14:   
  398500  XT-PIC  ide0 
  NMI:  
  0 
  ERR: 
  0I've tried modifying parameters in the bios and I didn't managed to 
  change the irq.Does anyone have a machine like mine?Have anyone 
  changed the irq in order to not sharing irq's? Thanks in 
  advance.
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RE: [Asterisk-Users] Web Admin

[Kerry Garrison]

You could install any number of interfaces but it does not 
come with one.
 
Kerry 
GarrisonDirector of Technical ServicesTech Data 
Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Francisco 
  SalinasSent: Wednesday, May 10, 2006 10:05 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Web 
  Admin
  
  
   
  I am 
  planning to deploy Asterisk Business edition. Does this edition have a web 
  module administration?
   
  Thanks
   
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RE: [Asterisk-Users] voipjet down?

[Kerry Garrison]

You always recommend CEO when ever anyone asks about a service provider when
nobody has asked about wholesale. Very very few people are interested in
wholesale pricing. If you are a consultant you want to recommend a company
to your clients that is the best fit for their needs. If you are an end user
you also do not care about wholesale pricing. So yes, everytime someone asks
for a recommendation you throw this incredibly expensive provider out there
and if anyone points out the fact that they are the single most expensive
provider available you turn around and say "but that's not their wholesale
price". Well, it IS the price they advertise on their website, and in fact,
it is the ONLY price advertised on their site. So you just keep extolling
the virtues of them and I will point out that it is not a good
recommendation if they are looking for a cost effective voip service
provider. Fair enough?

Secondly, read your own message, you recommend voipjet for both situations
when everyone is having issues with them. 

-Kerry

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Wednesday, May 10, 2006 11:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] voipjet down?
> 
> I'm not in bed with CEO.   All I do is use them... and want to see
> them get more customers.   Sorry if I sounded upset... it just seems
> like Kerry never reads the e-mail every time this issue comes up.
> 
> On 5/10/06, Jay Milk <[EMAIL PROTECTED]> wrote:
> > Whoa, kid, do you have a behavioral problem?  No reason to 
> get nasty.
> > You're only confirming the long-held suspicion that you're 
> in bed with 
> > calleveryone.  Friendly references are one thing, but 
> defending it at 
> > the expense of courtesy, that's suspicious.
> >
> > I don't think their customer service is all that top-notch 
> anyway.  I 
> > emailed them a pre-sales question on a weekday, and didn't get a 
> > response until almost 72 hours later.  Maybe their customer 
> service is 
> > better than their sales team, but in my experience, sales are 
> > generally the best staffed group in any service company.
> >
> > Lastly, the customer service quality of my termination provider is 
> > secondary.  A business that relies on outgoing calls wouldn't even 
> > touch IP termination for those essential calls; they'd get a PRI.  
> > Everyone else can wait a few minutes while the phone system 
> switches 
> > providers if one goes down.
> >
> > Matt wrote:
> > > Kerry,
> > > Do you have a reading problem?   Both times that I have 
> tried to help
> > > people out by suggesting a company I have personally used 
> and have had
> > > good luck with, you reply and say that the rates are 
> horrible.   If
> > > you would read my e-mails you would see that the 3.9 cents is NOT 
> > > for wholesale termination.
> > >
> > > If you want someone would will give cheap termination to 
> end users, 
> > > go use voipjet or whatever you want.
> > >
> > > If, on the other hand, you want some reliable cheap wholesale 
> > > termination, go check out voipjet.
> > >
> > > On 5/10/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> > >> I cant imagine anyone using voipjet as their only or 
> main provider.
> > >> And I'll
> > >> say again, 3.9 cents for an ITSP is the most expensive I 
> have found.
> > >> Business grade termination is typically much less than that with 
> > >> top notch companies like https://www.nexvortex.com/ at 2.5c.
> >
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> >
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RE: [Asterisk-Users] voipjet down?

[Kerry Garrison]

I cant imagine anyone using voipjet as their only or main provider. And I'll
say again, 3.9 cents for an ITSP is the most expensive I have found.
Business grade termination is typically much less than that with top notch
companies like https://www.nexvortex.com/ at 2.5c.
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Wednesday, May 10, 2006 5:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion; 
> Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-Users] voipjet down?
> 
> And again I'll say... calleveryone.com for all your RELIABLE 
> termination needs.  And again... don't go by the rates on the page...
> those are the end-user rates... call them for wholesale 
> rates.. they will be competitive to voipjet, and you get 
> phone support and quick response time.  Come on guys... if 
> you are still using VoipJet, you don't care about your 
> companies termination.
> 
> On 5/9/06, Wes Baehr <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Even stranger is when calls (to the same server) work from one 
> > asterisk server & account, but fail from another asterisk 
> server & account.
> > Sometimes changing the server helps, sometimes it doesn't.
> >
> >
> >
> >
> > Wes Baehr
> >
> > Ability Business Computing, Ltd.
> >
> > Office:  330.882.0455 x25
> >
> > Cell:  330.882.0455 x35
> >
> > Fax:  330.882.0455
> >
> >
> > [EMAIL PROTECTED]
> >
> >
> >
> >  
> >
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
> > Garrison
> >  Sent: Tuesday, May 09, 2006 4:21 PM
> >  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >  Subject: RE: [Asterisk-Users] voipjet down?
> >
> >
> >
> >
> > I havebent been able to call out in weeks and nobody 
> returns emails to 
> > [EMAIL PROTECTED]
> >
> >
> >
> >  
> >
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf 
> Of Julius 
> > Barber :: GringoTel.com
> >  Sent: Tuesday, May 09, 2006 12:40 PM
> >  To: asterisk-users@lists.digium.com
> >  Subject: [Asterisk-Users] voipjet down?
> >
> > Somebody know if they are down? Let me know,
> >
> >
> >
> >
> > Julius C. Barber
> >  [EMAIL PROTECTED]
> >  www.GringoTel.com
> >  Tel. USA: 1-408-705-1189
> >  GringoTel - ahorre en sus llamadas internacionales.
> >
> >
> >
> >
> >
> >
> > --
> >  No virus found in this incoming message.
> >  Checked by AVG Free Edition.
> >  Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 
> > 5/8/2006
> >
> >
> >
> >
> >
> > --
> >  No virus found in this outgoing message.
> >  Checked by AVG Free Edition.
> >  Version: 7.1.392 / Virus Database: 268.5.5/334 - Release Date: 
> > 5/8/2006
> >
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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RE: [Asterisk-Users] voipjet down?

[Kerry Garrison]

I havebent been able to call out in weeks and nobody 
returns emails to [EMAIL PROTECTED] 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Julius 
  Barber :: GringoTel.comSent: Tuesday, May 09, 2006 12:40 
  PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] voipjet down?
  
  
  Somebody know if they are down? 
  Let me know,
   
  
  Julius C. Barber[EMAIL PROTECTED]www.GringoTel.comTel. USA: 
  1-408-705-1189GringoTel - ahorre en sus llamadas 
  internacionales.
   
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RE: [Asterisk-Users] asterisk management interface

[Kerry Garrison]

[EMAIL PROTECTED] is an ISO image that installed CentOS, Asterisk, FreePBX, and
some other tools. FreePBX is just the web interface.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Zach A
> Sent: Tuesday, May 09, 2006 12:38 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] asterisk management interface
> 
> How different is FreePBX from Asterisk @ Home?
> 
> Zach
> 
> 
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RE: [Asterisk-Users] asterisk management interface

[Kerry Garrison]

Those are reasons for WANTING to create your own, he specifically said "I
HAVE to make my own" and I wanted to know why he HAS TO create his own when
there are fantastics products already available. There is a huge difference
in saying "I would like to create my own" and "I have to create my own". I
totally understand the 'want', I "want" something that is different and
don't the way I want but I don't "need" to right now.
-Kerry


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Senad Jordanovic
> Sent: Monday, May 08, 2006 2:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] asterisk management interface
> 
> [EMAIL PROTECTED] wrote:
> > http://www.freepbx.org
> >
> > Why would you need to create your own?
> 
> Many reasons:
> 
> 1. not relying on already busy open source developers 2. 
> creating something that you can possibly offer as your own 
> commercial offering 3. have it designed exactly they way you 
> want it from ground up 4. have a lot fun with it (and 
> headaches :) ) etc...
> 
> It is a long road though.
> We started PBXware in 2003 and there are still many features 
> we wish to implement.
> 
> 
> Senad
> 
> 
> 
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RE: [Asterisk-Users] asterisk management interface

[Kerry Garrison]

http://www.freepbx.org

Why would you need to create your own?

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> moona ather
> Sent: Monday, May 08, 2006 12:01 AM
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] asterisk management interface
> 
> As I know only php and no other langugae like perl or any 
> other... most of the links to such applications i have seen 
> on voip.org site made in php are removed or are inactive. Can 
> you tell me of any such application that i can use or make my 
> own using that made only in php and serving my pupose?
> thanx!
> 
> 
> >From: "Kerry Garrison" <[EMAIL PROTECTED]>
> >Reply-To: Asterisk Users Mailing List - Non-Commercial 
> >Discussion
> >To: "'Asterisk Users Mailing List - Non-Commercial 
> >Discussion'"
> >Subject: RE: [Asterisk-Users] asterisk management interface
> >Date: Sun, 7 May 2006 23:45:58 -0700
> >
> >Why make a brand new?
> >
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On 
> Behalf Of moona 
> > > ather
> > > Sent: Sunday, May 07, 2006 11:36 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: [Asterisk-Users] asterisk management interface
> > >
> > > Hi,
> > > I have to make a web-based management interface of configuring 
> > > asterisk
> > > i wanted to know if it is as simple as reading the .conf 
> files and 
> > > searching for the required section in the file and adding 
> users etc. 
> > > or there are other steps involved too?? As I have seen many such 
> > > built codes on this site and found lots of code... kindly tell me 
> > > how complex it is and how many other steps are involved in making 
> > > this interface as i am new in this.
> > > Emmo.
> > >
> > > _
> > > Don't just search. Find. Check out the new MSN Search!
> > > http://search.msn.click-url.com/go/onm00200636ave/direct/01/
> > >
> > > ___
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> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
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> _
> Express yourself instantly with MSN Messenger! Download today 
> it's FREE! 
> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
> 
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RE: [Asterisk-Users] asterisk management interface

[Kerry Garrison]

Why make a brand new?
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> moona ather
> Sent: Sunday, May 07, 2006 11:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] asterisk management interface
> 
> Hi,
> I have to make a web-based management interface of 
> configuring asterisk 
> i wanted to know if it is as simple as reading the .conf 
> files and searching for the required section in the file and 
> adding users etc. or there are other steps involved too?? As 
> I have seen many such built codes on this site and found lots 
> of code... kindly tell me how complex it is and how many 
> other steps are involved in making this interface as i am new in this.
> Emmo.
> 
> _
> Don't just search. Find. Check out the new MSN Search! 
> http://search.msn.click-url.com/go/onm00200636ave/direct/01/
> 
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RE: [Asterisk-Users] asterisk hardware

[Kerry Garrison]

He asked about hard phones not soft phones. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steve Totaro
> Sent: Sunday, May 07, 2006 12:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] asterisk hardware
> 
> Give idefisk a try.  It works very well for me, its free, and 
> does not crash all the time like Cubix (formerly Firefly).
> 
> 
> Tofik Suleymanov wrote:
> > Hello folks,
> >
> > anyone using hardware IAX phones with asterisk ?
> > I've googled on this issue and found several hardware phones which 
> > support IAX protocol, but before paying money I'd like to know more 
> > about what people experiencing with them.
> >
> >
> > Thank you,
> > Tofik Suleymanov
> >
> 
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[Asterisk-Users] CW options not changing

[Kerry Garrison]

I have a weird issue 
with a system, running freepbx in devices and users mode (same as dozens of 
systems) but this one when you hit *70 and get "call waiting activated" it is 
not storing the setting in the database. I can manually set CW 900 ENABLED but 
then *71 does not disable it. Any suggestions?
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

[Kerry Garrison]

Go into the BIOS, disable every possible device such as floppy controller,
usb, serial, parallel, etc. If that doesn't work, move card to another slot.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Giorgio Incantalupo
> Sent: Friday, May 05, 2006 1:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with IRQ conflict between 
> wct2xxp and eth0
> 
> Hi Phil,
> may sound a stupid advice buthave you tried to change PCI 
> slots on your server?
> 
> Giorgio
> 
> Phil Menico wrote:
> >
> > I have a conflict problem with the eth0 card and wct2xxp 
> digium board. 
> > The PRI can receive calls but my network connection is gone.
> >
> > When I "cat /proc/interrupts" I get the following:
> >
> > 1 ..
> >
> > 1 ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > 169 0 IO-APIC-level wct2xxp, eth0
> >
> > ..
> >
> > etc.
> >
> > even before I "modprobe wct2xxp"
> >
> > After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat 
> > /proc/interrupts"
> >
> > I then get:
> >
> >  
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > 169 118489 IO-APIC-level wct2xxp, eth0
> >
> > 201 118497 IO-APIC-level wctdm
> >
> > ..
> >
> > etc
> >
> >  
> >
> > How can I force the wct2xxp to load on a separate IRQ? I 
> tried moving 
> > the eth0 to IRQ 10 but could not.
> >
> > Any ideas?
> >
> >  
> >  
> >
> > Thank you.
> >
> > _*/Phil Menico/*_
> >
> > XTEND Communications
> > 171 Madison Avenue, New York, NY 10016
> > 212-951-7632 (Office)
> > 212-951-7683 (Fax)
> > www.xtend.com
> >
> >  
> >
> > 
> --
> > --
> >
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> >   
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RE: [Asterisk-Users] Voipjet Problem?

[Kerry Garrison]

Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service may be stellar but when clients are actually trying to save money,
that's a damned hard sell.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Thursday, May 04, 2006 8:18 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voipjet Problem?
> 
> Just wanted to add my 2 cents.  We were with voipjet, and do still use
> them for occassional backup.However, their lack of personal
> service and inability to get ahold of someone drove us away.After
> several total blackouts (like what happened yesterday), and 
> no responce we finally put out an SOS on the asterisk mailing 
> list.  Of course there were several responces from companies 
> trying to solicit us. but the one that caught our 
> attention was calleveryone.com  
> So far we have been rock-solid-happy with them.   We've had a few
> small bumps along the road.   For instance, once there was a router
> along our path to them that was dropping packets, but this was quickly
> resolved.   Additionally, they've worked with us on the phone to
> resolve audio problems, and diagnose carrier issues.   If I have a
> problem, I rest assured that I can call someone, or page 
> someone if the situation is severe enough, and get ahold of a 
> human at any hour
> of the evening.   Not so with VoipJet.   I don't want to bad mouth
> VoipJet, their service is decent... but definately not acceptable for
> a carrier grade level.   I'm not affiiliated with calleveryone in any
> way other then a very happy and satisfied customer, and would highly
> recommend them to you.   If you are a wholesole buyer of minutes, talk
> to them, don't just take their prices on the main page... 
> those are for residential and regular customers.  Their 
> prices are very comparable to voipjet, and the service is miles ahead.
> 
> On 5/3/06, Matt <[EMAIL PROTECTED]> wrote:
> > Yup... I think they died... this is why I stopped using 
> them except as
> > my backup.   It seems 64.34.45.100  is working ok as of right now.
> > It wouldn't be so bad if they had a number you could call 
> for support!
> >  HERE THAT JOHN?   You need a phone number if you want to "play with
> > the big dogs".
> >
> > On 5/3/06, Mark Hulber <[EMAIL PROTECTED]> wrote:
> > > I started to have a problem today that all my calls 
> through voipjet 
> > > result in just timing out after my assigned timeout 
> period.  I tried 
> > > multiple of their servers with the same problem.  Anyone 
> else having 
> > > a problem?  I am running:
> > >
> > > Asterisk SVN-branch-1.2-r24381M built by root @ 
> asterisk.hulber.com 
> > > on a
> > > i686 running Linux on 2006-05-03 14:14:07 UTC
> > >
> > > I can connect with other IAX providers.
> > >
> > > MARK.
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
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RE: [Asterisk-Users] Simple Dell Computers

[Kerry Garrison]

Most of the Dells will work fine with some minor workarounds. First off, go
into the BIOS and disable every possible device (USB, Floppy controller
serial, parallel, etc). Then if the card does not work correctly, move it to
a different slot. With most of the lower end Dells you will find that the
card will only function properly in one of the three PCI slots. If you get a
motherboard that has more than 3 PCI slots your chances of success are
dramatically higher.

Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Dovid Bender
> Sent: Wednesday, May 03, 2006 5:10 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Simple Dell Computers
> 
> Hello List,
> I know this has been brought up many times but I wanted to 
> know if anyone had any expirience in the following. I setting 
> up several voice mail systems.
> Each one is going to have a TDM400P. Two FXO for people to 
> leave messages and two FXS for POTS phones so people can 
> listen. Anyone know if there are any simple specific dell 
> models that will handle this without a problem ?
> 
> Thanks.
> 
> Dovid
> 
> __
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> around http://mail.yahoo.com 
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RE: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset

[Kerry Garrison]

Are you seriously trying to run 4 cards in one system? The odds of getting
that working are about the odds of Angelina Jolie showing up on my doorstep
ready to whisk me off tobut I digress...you will have serious interrupt
issues trying to get 4 cardss working in one system. I am surprised that you
would fork for a PRI card but use cheap winmodems for analog lines. You will
have much better luck tossing the "x100p" cards and using either SPA-3000's,
a TDM400, or a Mediatrix 1204.

Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
> Sent: Tuesday, May 02, 2006 7:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Need help configuring TE100P and 3 
> X100P clonewith MD3200 chipset
> 
>  
> I can either get the TE100P working or the 3 X100P clones 
> working, but never both. I have the TE100P connected to a 
> channel bank, and X100P clones to lines from the phone company. 
> 
> 
> This is my zaptel.conf
> 
>  span=1,1,0,d4,ami
>  fxsks=1-24
>  loadzone=us
> 
>  fxols=25-27
>  loadzone=us
> 
> I then do
> 
> [EMAIL PROTECTED] root]# modprobe zaptel
> [EMAIL PROTECTED] root]# modprobe  wcte11xp
> ZT_CHANCONFIG failed on channel 25: No such device or address (6)
> /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed
> /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed 
> [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on 
> channel 25: Invalid argument (22) Did you forget that FXS 
> interfaces are configured with FXO signalling and that FXO 
> interfaces use FXS signalling?
> /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed
> /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed 
> [EMAIL PROTECTED] root]# 
> 
> What's wrong with configuration?
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RE: [Asterisk-Users] FreePBX in production?

[Kerry Garrison]

> There are still some basic things missing (for example if you
don't use 
> voicemail it is not possible to set a destination for the call if
not 
> answered, you have to create a ring group for each extension to
work 
> around it, this is a major issue)

Remco - take a look at the Follow Me module I added. It is basically
a presonal ring group for each extension. If you want to do the above, just
define the Follow-Me settings to ring your own extension (or more if you
want) and then choose any destination you want. It effectively does 'creat a
ring group for each extensions' that wants one, but it does it in such a way
as to be separate and work side by side with normal ringgroups, and there is
a direct link between it and the extension (or user) so that navigation is
very easy as you can bounce back and forth with a single mouse click.
 

Many people have talked about limitations of freePBX and how you cant do
custom things. Both Phillip and I attacked the follow-me function this week
with his using personal ring groups and mine using personal call queues (see
article at http://voipspeak.net) to simulate the functionaly of the locate
function from CallManager. Both solutions used only freePBX functionality to
acocmplish two relativly complex tasks that many people have been struggling
to create with just the config files. That speaks to the flexibility of
using the system. Sure there are bound to be limitations that may prevent
some type of functionality, no system is perfect, but it does cover far more
than just a few small businesses.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



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RE: [Asterisk-Users] Softphone ready to go installed on USB flashdrive

[Kerry Garrison]

The current versions of IDEFISK use a Windows installer, 
wether it is required or not now I dont know.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
  ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Softphone ready to go installed on USB 
  flashdrive
  I do this with the windows version of idefisk from Asteriskguru.com. The configuration is 
  stored in the dir with the program and dll. I have actually configured it and 
  emailed it to users. There is no installer and a simple shortcut or autoplay 
  menu should take care of the rest. 
  On 5/1/06, Time 
  Bandit <[EMAIL PROTECTED]> 
  wrote:
  > 
How can I install a softphone on my USB flash drive like Xlite and 
have> it ready to go when I plug  it in at any Windows XP 
computer?> (Same for a Linux softphone, both on one USB flash 
drive).I believe Dan's softphone is suitable for this. See http://www.laser.com/dante/diax/diax.htmlActually, 
I should do that with my softphone instead of using the registry 
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RE: [Asterisk-Users] FreePBX in production?

[Kerry Garrison]

You can already do that. You ca specify different access to different users
with the Administrators module.
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Monday, May 01, 2006 6:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] FreePBX in production?
> 
> Time Bandit wrote:
> >> > You could, but it'll get overwritten by any FreePBX upgrades. The
> >> *.conf
> >> > and *_additional.conf files are controlled by FreePBX and can be 
> >> > overwritten.
> >>
> >> I thought I should clarify this statement: I meant that 
> FreePBX could 
> >> overwrite both the *.conf and the *_additional.conf files. You are 
> >> strongly advised NOT to edit either of those types of files. All 
> >> editing should be restricted to the *_custom.conf files.
> > Well, I've modified *.conf files and I never had AMP (FreePBX) 
> > overwrite them. An upgrade would most certainly overwrite 
> them but not 
> > normal usage. I may be wrong, but if you upgrade, the *_custom.conf 
> > files will probably get overwritten also, so you better backup them 
> > before.
> 
> Let's see if I can summarize various recent postings relative 
> to the broader topic of whether FreePBX/AAH is production-ready.
> 
> Seems the general consensus is that AAH and/or FreePBX is 
> considered production ready if the functionality embedded in 
> AMP (primarily) happens to fit the specific small business 
> requirements. Anything outside of the basic functionality is 
> limited primarily by the lack of technical documentation, the 
> undocumented logic behind magically creating dialplan 
> entries, and limitations associated with AMP interfaces to 
> various channels such as those typically defined in zapata.conf, etc.
> 
> It would almost appear as though the user interface should be 
> broken into two components: 1) a simplified interface for 
> non-technical users that are responsible for 
> adds/moves/changes, and, 2) a second interface to define 
> business-specific items such as defining certain interfaces 
> (eg, zap channels), contexts, dialplans, etc. Many of those 
> items defined in #2 would probably become drop-down 
> selections for the user interface in #1.
> 
> Thoughts?
> 
> R.
> 
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RE: [Asterisk-Users] queues

[Kerry Garrison]

This is not the right place for help with AAH. Use the AAH forum at sf.net.

If it is just hanging up on users, it is not configured properly.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Patrick Siglin
> Sent: Sunday, April 30, 2006 8:25 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] queues
> 
> I am not understanding how queues are supposed to work. I am 
> using [EMAIL PROTECTED] and configured a queue in AMP. I have 
> also set my static extensions in the queue. If I set up the 
> system to put people in the queue on incoming it just hangs 
> up on them. If I try to log in as an agent it says I am 
> logged in and then disconnects. If I do a show agents it says 
> I'm not logged in. I looked at some samples but not quite 
> getting it. My queue is 100 and my two extensions are 200 and 201.
> 
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RE: [Asterisk-Users] FreePBX in production?

[Kerry Garrison]

Its just you.

There is much more flexibility on handling incoming pstn lines than there
was in the last version of AMP 

If you like manually creating config files with custom settings for each
user, then a GUI is not for you.  I have several clients using freePBX
because it is easier to maintain some of the features they wanted this way
than dealing with the config files. 

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Sunday, April 30, 2006 2:20 PM
> To: Asterisk Users-List
> Subject: [Asterisk-Users] FreePBX in production?
> 
> Has anyone attempted to use FreePBX for a business in production mode?
> 
> Initial take is there are lots of things scripted but a lot 
> of limitations in terms of supporting basic business 
> functions. Inability (or lack of flexibility) is handling 
> multiple incoming pstn lines, dialplan limitations, poor/no 
> documentation, etc, to mention a few.
> 
> Maybe its just me, but it appears its no where near usable 
> even with the latest beta1 code.
> 
> Is it just me or what?
> 
> Rich
> 
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RE: [Asterisk-Users] RE: Asterisk is stripping my area code

[Kerry Garrison]

Do you have 9 as a prefix in the trunk? It is actually 
ADDING a 9 to the phone number before it dials.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] RE: Asterisk is stripping my area code
  No, I'm just dialing 7707190239.  When I tried it with a 1, it 
  gave me the same result, a nice lady telling me "when making a local call you 
  must first dial the areacode" or words to that effect.  >From 
  the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c: 
  Function result is '"" <7707190069>'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing NoOp("SIP/200-fa0b", "CallerID 
  set to "" <7707190069>") in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "GROUP()=OUT_1") in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function 
  result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is 
  '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing GotoIf("SIP/200-fa0b", "0?108") in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "DIAL_NUMBER=17707190239") in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "DIAL_TRUNK=1") in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing AGI("SIP/200-fa0b", 
  "fixlocalprefix") in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242] 
  logger.c:   fixlocalprefix: Removed prefix. New number: 
  7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: 
  -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 
  VERBOSE[4242] logger.c: -- Executing 
  Set("SIP/200-fa0b", "OUTNUM=97707190239") in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "custom=ZAP/1") in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ 
  result is '0'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing GotoIf("SIP/200-fa0b", "0?16") 
  in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any 
  branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing Dial("SIP/200-fa0b", "ZAP/1/97707190239|120|W") in new stackApr 
  30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04 
  DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Called 1/97707190239Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 
  (index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 
  1So what should I have in my dialing plan to let me dial 7707190239 
  and have it use that exact number?  Or do I have to dial 9 
  first?Thanks, Jim.
  On 4/30/06, Kerry 
  Garrison <[EMAIL PROTECTED]> 
  wrote:
  

Are you 
dialing 9 first? It is showing that the digits you dialed 
are:
 
9-770-719-0239
Using your dialplan you 
should be dialing 1-770-719-0239
 


Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949) 502-7819 x200 - 
[EMAIL PROTECTED] 
http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Jim LynchSent: Sunday, April 30, 2006 10:10 
  AMTo: Asterisk-Users@lists.digium.comSubject: 
  [Asterisk-Users] RE: Asterisk is stripping my area 
  code

I don't know if this helps, from the log.Jim.Apr 
30 12:58:55 VERBOSE[4225] logger.c: -- Executing 
Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new 
stack   
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
12:58:55 DEBUG[4225] chan_zap.c: Deferring 
dialing...    
Apr 30 12:58:55 VERBOSE[4225] logger.c: -- 
Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 
19, channel 
1 
 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
Complete(12) on channel 1 (index 
0)    

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RE: [Asterisk-Users] RE: Asterisk is stripping my area code

[Kerry Garrison]

Are you dialing 9 first? It is showing that the digits you 
dialed are:
 
9-770-719-0239
Using your dialplan you should be dialing 
1-770-719-0239
 


Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:10 AMTo: 
  Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE: 
  Asterisk is stripping my area code
  I don't know if this helps, from the log.Jim.Apr 30 
  12:58:55 VERBOSE[4225] logger.c: -- Executing 
  Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new 
  stack   
  Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
  12:58:55 DEBUG[4225] chan_zap.c: Deferring 
  dialing...    
  Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 
  1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, 
  channel 
  1 
   Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
  Complete(12) on channel 1 (index 
  0)    
  
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RE: [Asterisk-Users] RE: Install/Upgrade

[Kerry Garrison]

upgrading from what version?

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dave 
  MorrowSent: Saturday, April 29, 2006 6:11 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] RE: Install/Upgrade
  
   
  
  Hi all, I was just 
  wondering if anyone knows of any gotchas with respect to upgrading 
  Asterisk to the latest 1.2.7 ?
   
  Is the procedure 
  the same?  Config files remain intact?  Just untar/make 
  install?
   
  David Morrow
  Technical Systems Lead
  Autodata Solutions 
Company
  [EMAIL PROTECTED]
  http://www.autodatasolutions.com
   
  Tel: (519) 963-3020
  Fax: (519) 451-6615
   
  < Lead, follow or get out of 
  the way! >
   
  
  This message has originated from Autodata Solutions. The attached material 
  is the Confidential and Proprietary Information of Autodata Solutions. This 
  email and any files transmitted with it are confidential and intended solely 
  for the use of the individual or entity to whom they are addressed. If you 
  have received this email in error please delete this message and notify the 
  Autodata system administrator at [EMAIL PROTECTED] 
   
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[Asterisk-Users] Locate Me Function with freePBX

[Kerry Garrison]

The client's needs are the mother of invention. We 
have a client that currently uses a Cisco Call Manager and one of the features 
they love was the Locate-Me function (or follow-me, or find-me, whatever you 
want to call it) which basically rings their desk phone a few times then plays a 
short message, and then rings their other remote phones and cell phones. The 
customer wants this same functionalty from an Asterisk system that we will be 
building running freePBX.  
 
It took me a while to think about how to implement 
this without mucking around with the config files but the end result is a fairly 
simple solution that enables the use to turn on/off the function at 
will.
 
For the complete article on how to implement this 
feature, go to:http://voipspeak.net/index.php?option=com_content&task=view&id=72
 


Kerry 
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

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RE: [Asterisk-Users] FW: NuFone Update: DIDs

[Kerry Garrison]

AMEN!!
 
Any consultant that DOESNT take this into consideration 
should stick to installing Windows and calling themselves an IT 
"Expert".
 
You can screw up someone's network, mess up a 
workstation, hose their email, but you break someone's telephone service there 
will be hell to pay.
 
Kerry 
GarrisonDirector of Technical ServicesTech Data 
Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Lacy 
  Moore - AspendoraSent: Friday, April 28, 2006 4:39 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] FW: NuFone Update: DIDs
  
  Exactly why I chose to go with a PRI for business use.  There is 
  something to be said for the stability of a telco, even if it's not SBC (or 
  AT&T now).  In some cases, government interference is good.  How 
  many businesses can survive a loss of their phone number?  I know the 
  ones I deal with cannot. 
   
  This is something we need to take into consideration as Asterisk users or 
  consultants.  We need to look at the whole picture, not just a short term 
  savings.  Can your business/clients survive without listings in directory 
  assistance or the phone books?  Can they survive if they have to change 
  numbers due to their Voip provider losing a contract?  Some can, most 
  can't.  I looked into using Voip.  Technically, it seemed like a 
  good solution.  I just don't trust it in the long run.  I know that 
  by using a telco, I will have access to my phone numbers.  With Voip 
  providers, who controls the numbers?  
   
  I think this is something a lot of people fail to take into 
  consideration.
   
  On 4/28/06, Chris Mason 
  (Lists) <[EMAIL PROTECTED]> 
  wrote: 
  I 
would be very wary, as VOIP providers feel no responsibility to 
thecustomer and will not bother to tell us they might not be around next 
week. Once bitten...--Chris MasonNetConcepts(264) 
497-5670 Fax: (264) 497-8463Int:  (305) 704-7249 Fax: 
(815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This 
message has been scanned for viruses anddangerous content by 
MailScanner, and isbelieved to be 
clean.___ 
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listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. 
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RE: [Asterisk-Users] USB conference phone

[Kerry Garrison]

I use a softphone 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> John covici
> Sent: Thursday, April 27, 2006 9:11 PM
> To: Steve Feinstein
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] USB conference phone
> 
> OK, assuming the usbaudio sees the conference phone and can 
> work it, how would you write an extension to ring that?
> 
> on Thursday 04/27/2006 Steve Feinstein([EMAIL PROTECTED]) 
> wrote  > It's a standard USB audio device.  While I haven't 
> tried it, I'm pretty  > sure the Linux USB audio driver will 
> probably see it.
>  >
>  > -Steve
>  >
>  > John covici wrote:
>  > > Any way to use this on a Linux box so I could use this 
> with asterisk?
>  > > I have a windows box on the same network, but how would 
> I get asterisk  > > to see such a thing?
>  > >
>  > > Thanks.
>  > >
>  > > on Wednesday 04/26/2006 Steve 
> Feinstein([EMAIL PROTECTED]) wrote  > >  > 
> http://www.iogear.com/main.php?loc=product&Item=GPH100U
>  > >  >
>  > >  > I've got a couple of these, they're $40 and there's a 
> $20 rebate going  > >  > on now. For that price it's pretty 
> amazing. Plug and play, no drivers  > >  > required. Quality 
> very good, it does echo cancellation and noise  > >  > 
> reduction. I only wish it had a mute button.
>  > >  >
>  > >  > BTW: I don't have any affiliation with ioGear other 
> than I like this  > >  > product.
>  > >  >
>  > >  >
>  > >  > Jim Houser wrote:
>  > >  > > Personal preference. I'm not a big headset guy.
>  > >  > > The real point of my reply was to say how impressed 
> I am with USB talk  > >  > > quality when compared to a 
> hardphone on Asterisk or our Avaya  > >  > > Communications 
> Manager. Like my wife says, I guess I'm not being  > >  > > 
> clear... :)  > >  > >  > >  > > 
> --
> --
>  > >  > > *From:* [EMAIL PROTECTED]
>  > >  > > [mailto:[EMAIL PROTECTED] 
> *On Behalf Of *Dean  > >  > > Collins  > >  > > *Sent:* 
> Wednesday, April 26, 2006 10:24 AM  > >  > > *To:* Asterisk 
> Users Mailing List - Non-Commercial Discussion  > >  > > 
> *Subject:* RE: [Asterisk-Users] USB conference phone  > >  > 
> >  > >  > > Kerry, do you actually own one? Have you used it 
> for long? What are  > >  > > you using it for?
>  > >  > >
>  > >  > > (jim ? personally I cant see the point of using 
> your phone when I have  > >  > > a very good quality headset 
> and mic.).
>  > >  > >
>  > >  > > Dean
>  > >  > >
>  > >  > > 
> --
> --
>  > >  > >
>  > >  > > *From:* [EMAIL PROTECTED]
>  > >  > > [mailto:[EMAIL PROTECTED] 
> *On Behalf Of *Kerry  > >  > > Garrison  > >  > > *Sent:* 
> Wednesday, 26 April 2006 10:36 AM  > >  > > *To:* 'Asterisk 
> Users Mailing List - Non-Commercial Discussion'
>  > >  > > *Subject:* RE: [Asterisk-Users] USB conference 
> phone  > >  > >  > >  > > This is an excellent USB 
> speakerphone  > >  > >  > >  > > 
> http://voipspeak.net/index.php?option=com_content&task=view&id
=39&Itemid=27
>  > >  > > 
> <http://voipspeak.net/index.php?option=com_content&task=view&i
d=39&Itemid=27>
>  > >  > >
>  > >  > > 
> --
> --
>  > >  > >
>  > >  > > *From:* [EMAIL PROTECTED]
>  > >  > > 
> [mailto:[EMAIL PROTECTED] *On Behalf Of
>  > >  > > *Jim Houser
>  > >  > > *Sent:* Wednesday, April 26, 2006 6:26 AM
>  > >  > > *To:* 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'
>  > >  > > *Subject:* RE: [Asterisk-Users] USB conference phone
>  > >  > >
>  > >  > > I don't know about this phone but I can tell 
> you I have a vendor
>  > >  > > that will only talk to me via Skype so I purchas

RE: [Asterisk-Users] Looking for input on which way to gowith smallbusiness setup

[Kerry Garrison]

What would the difference be with using IP phones or ATA's in that case? You
are still talking about a network device. Even with 50-60 stations installs
you are highly unlikely to run into a need for QoS internally.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jean-Michel Hiver
> Sent: Thursday, April 27, 2006 1:41 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Looking for input on which way 
> to gowith smallbusiness setup
> 
> Kerry Garrison a écrit :
> 
> >You will kick yourself up and down the block for not using 
> IP phones in 
> >the end. What are you going to spend? $65 for an ata and $25 for a 
> >phone? Spend the extra money and get SNOM or Linksys phones. 
> You then 
> >need to figure out how to get 12 analog lines into Asterisk. Using 3 
> >TDM400's is not really an option as you will spend countless hours 
> >trying to figure out interrupt issues.  The next best option is 3 
> >Mediatrix 1204 Gateways, this will set you back about $1,800 
> and will 
> >be maxed out. The best option, sticking with analog lines that is, 
> >would be a Rhino CB-24 FXO Channel Bank connected to a Rhino 
> R1T1 card 
> >in the asterisk server. This bundle will hit you for about 
> $2,000 but will only be half full.
> >  
> >
> True, but using IP Phones means you need to do proper QoS at 
> the level of your LAN or install a second LAN (which kinda 
> defeats the purpose) as otherwise a network card going 
> bananas could kill your phone system. So you need to get a 
> proper QoS enabled switch, which is quite expensive too...
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RE: [Asterisk-Users] Polycom NTP issue

[Kerry Garrison]

I changed the DHCP overrides to 1, restarted the phone, now it showing the
right day but it is 4 hours and 29 minutes fast. Arrrgh.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Thursday, April 27, 2006 1:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom NTP issue
> 
> I don't have a polycomm manual handy, but I think I'd change 
> the overrideDHCP parameter to "1" and test. You are 
> apparently in the PST timezone?
> 
> If that doesn't do it, my next step would be to use ethereal 
> to capture one of the ntp request/response pkts and analyze 
> the content. If that looks okay, then something in the phone 
> isn't right.
> 
> 
> Kerry Garrison wrote:
> > Here is the sip.cfg file
> > 
> >> tcpIpApp.sntp.address="192.168.10.50" 
> > tcpIpApp.sntp.address.overrideDHCP="0" 
> > tcpIpApp.sntp.gmtOffset="-25200" 
> > tcpIpApp.sntp.gmtOffset.overrideDHCP="0" 
> > tcpIpApp.sntp.daylightSavings.enable="1" 
> > tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
> > tcpIpApp.sntp.daylightSavings.start.month="4" 
> > tcpIpApp.sntp.daylightSavings.start.date="1" 
> > tcpIpApp.sntp.daylightSavings.start.time="2" 
> > tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
> > tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
> > tcpIpApp.sntp.daylightSavings.stop.month="10" 
> > tcpIpApp.sntp.daylightSavings.stop.date="1" 
> > tcpIpApp.sntp.daylightSavings.stop.time="2" 
> > tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
> > tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="1"/>
> > 
> > Using the same IP for the server on a Linksys SPA-941 everything is 
> > correct, using this configuration on the 501 shows 24 hours ahead.
> > 
> > -Kerry
> > 
> >  
> > 
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
> >> [mailto:[EMAIL PROTECTED] On Behalf Of Rich 
> >> Adamson
> >> Sent: Thursday, April 27, 2006 11:59 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [Asterisk-Users] Polycom NTP issue
> >>
> >> Kerry Garrison wrote:
> >>> I am ready to pull my hair out. I cannot seem to get the
> >> Polycoms to
> >>> read the time properly. Regardless of the server they are
> >> pointed to
> >>> our the offset, i am getting the correct time, but 24 hours
> >> ahead. So
> >>> for today it is showing Friday April 28 but with the
> >> correct time. Any clues?
> >>
> >> This is a copy/paste of the exact statements used on a IP600 phone:
> >>
> >>  >> tcpIpApp.sntp.address="134.84.84.84"
> >> tcpIpApp.sntp.gmtOffset="-21600" 
> >> tcpIpApp.sntp.daylightSavings.enable="1" 
> >> tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
> >> tcpIpApp.sntp.daylightSavings.start.month="4" 
> >> tcpIpApp.sntp.daylightSavings.start.date="1" 
> >> tcpIpApp.sntp.daylightSavings.start.time="2" 
> >> tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
> >> tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
> >> tcpIpApp.sntp.daylightSavings.stop.month="10" 
> >> tcpIpApp.sntp.daylightSavings.stop.date="1" 
> >> tcpIpApp.sntp.daylightSavings.stop.time="2" 
> >> tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
> >> tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="1"
> >>
> >> Are you sure you are setting the gmtOffset to the proper 
> number? The 
> >> above example is for CST, which is -6 hours (or -21600 
> seconds) from 
> >> GMT. It is also config'ed with resyncPeriod = 24 hours, 
> meaning the 
> >> clock is only sync'ed once per day.
> >>
> >> What are you using for the above?
> >>
> >>
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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > 
> > 
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RE: [Asterisk-Users] Looking for input on which way to go with smallbusiness setup

[Kerry Garrison]

You will kick yourself up and down the block for not using IP phones in the
end. What are you going to spend? $65 for an ata and $25 for a phone? Spend
the extra money and get SNOM or Linksys phones. You then need to figure out
how to get 12 analog lines into Asterisk. Using 3 TDM400's is not really an
option as you will spend countless hours trying to figure out interrupt
issues.  The next best option is 3 Mediatrix 1204 Gateways, this will set
you back about $1,800 and will be maxed out. The best option, sticking with
analog lines that is, would be a Rhino CB-24 FXO Channel Bank connected to a
Rhino R1T1 card in the asterisk server. This bundle will hit you for about
$2,000 but will only be half full.

However, you say you have 12 extensions and 12 lines? It is very rare to
have a 1:1 ratio. Usually for 12 people you will see 5-8 lines needed.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw
> Sent: Thursday, April 27, 2006 1:09 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Looking for input on which way to 
> go with smallbusiness setup
> 
> Hey guys!
> I'm the past week and a half, I have really learned a lot 
> from the mailing list and the wiki's posted online.
> 
> Now I have a question regarding different ways I can setup my 
> asterisk server for a small business with 12 extensions in 
> the office. Cost is a great concern, so I know cheap analog 
> phones at the desks is what we are looking at. My question 
> is, should I go do a fractional T1 for voice only, and the 
> get a Internet Service provider for their data needs, get 
> some sort of ATA and run the 12 analog phones to that OR go 
> with what SBC/AT&T (in my
> area) is offering, and do the "Integrated T1" option and have 
> just the 12 channels for voice, run some ATA between * and 
> the phones and use the 768 up and down for data? Anyone use 
> this "Integrated T1" with asterisk before?
> What hardware did you use?
> 
> Thanks for your input!
> 
> 
> Terrelle
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RE: [Asterisk-Users] Polycom NTP issue

[Kerry Garrison]

Here is the sip.cfg file

  

Using the same IP for the server on a Linksys SPA-941 everything is correct,
using this configuration on the 501 shows 24 hours ahead.

-Kerry

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Thursday, April 27, 2006 11:59 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom NTP issue
> 
> Kerry Garrison wrote:
> > I am ready to pull my hair out. I cannot seem to get the 
> Polycoms to 
> > read the time properly. Regardless of the server they are 
> pointed to 
> > our the offset, i am getting the correct time, but 24 hours 
> ahead. So 
> > for today it is showing Friday April 28 but with the 
> correct time. Any clues?
> 
> This is a copy/paste of the exact statements used on a IP600 phone:
> 
>  tcpIpApp.sntp.address="134.84.84.84"
> tcpIpApp.sntp.gmtOffset="-21600" 
> tcpIpApp.sntp.daylightSavings.enable="1" 
> tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" 
> tcpIpApp.sntp.daylightSavings.start.month="4" 
> tcpIpApp.sntp.daylightSavings.start.date="1" 
> tcpIpApp.sntp.daylightSavings.start.time="2" 
> tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" 
> tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" 
> tcpIpApp.sntp.daylightSavings.stop.month="10" 
> tcpIpApp.sntp.daylightSavings.stop.date="1" 
> tcpIpApp.sntp.daylightSavings.stop.time="2" 
> tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" 
> tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="1"
> 
> Are you sure you are setting the gmtOffset to the proper 
> number? The above example is for CST, which is -6 hours (or 
> -21600 seconds) from GMT. It is also config'ed with 
> resyncPeriod = 24 hours, meaning the clock is only sync'ed 
> once per day.
> 
> What are you using for the above?
> 
> 
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RE: [Asterisk-Users] Polycom NTP issue

[Kerry Garrison]

DNS is Windows 2003
Using the NTP server from CentOS 4.3
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Aaron Daniel
> Sent: Thursday, April 27, 2006 9:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom NTP issue
> 
> What dns server are you running?
> 
> On Thu, 27 Apr 2006, Kerry Garrison wrote:
> 
> > I am ready to pull my hair out. I cannot seem to get the 
> Polycoms to 
> > read the time properly. Regardless of the server they are 
> pointed to 
> > our the offset, i am getting the correct time, but 24 hours 
> ahead. So 
> > for today it is showing Friday April 28 but with the 
> correct time. Any clues?
> >
> > Kerry Garrison
> > Director of Technical Services
> > Tech Data Pros - Orange County's Mobile IT Service Provider
> > (949) 502-7819 x200 -  <mailto:[EMAIL PROTECTED]> 
> > [EMAIL PROTECTED] <http://www.techdatapros.com/> 
> > http://www.techdatapros.com
> >
> >
> 
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> [EMAIL PROTECTED]
> (936) 294-4198
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RE: [Asterisk-Users] Polycom NTP issue

[Kerry Garrison]

Polycom 501 
Firmware: 1.6.2.0041
Bootrom: 3.1.0.0269

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matt Florell
> Sent: Thursday, April 27, 2006 9:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom NTP issue
> 
> What Polycom phone model?
> 
> What firmware version?
> 
> What bootROM version?
> 
> Older versions of Polycom phones only worked with SNTP time 
> servers not NTP.
> 
> MATT---
> 
> On 4/27/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> >
> > I am ready to pull my hair out. I cannot seem to get the 
> Polycoms to 
> > read the time properly. Regardless of the server they are 
> pointed to 
> > our the offset, i am getting the correct time, but 24 hours 
> ahead. So 
> > for today it is showing Friday April 28 but with the 
> correct time. Any clues?
> >
> >  Kerry Garrison
> > Director of Technical Services
> > Tech Data Pros - Orange County's Mobile IT Service Provider
> > (949) 502-7819 x200 - [EMAIL PROTECTED] 
> > http://www.techdatapros.com
> >
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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[Asterisk-Users] Polycom NTP issue

[Kerry Garrison]

I am ready to pull 
my hair out. I cannot seem to get the Polycoms to read the time properly. 
Regardless of the server they are pointed to our the offset, i am getting the 
correct time, but 24 hours ahead. So for today it is showing Friday April 28 but 
with the correct time. Any clues?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Asterisk as a phone survey system

[Kerry Garrison]

Asterisk is simply a telephony toolkit, so the simple 
answer is yes, Asterisk can do this. Also, being a toolkit means there are a 
number of ways to accomplish it. You could right PERL, Python, TCL, C, PHP or 
numerous other types of scripts that can manage this for you. To see how to do 
some of the basic functions, you can look at some of the scripts at Nerd Vittles 
(http://nerdvittles.com). Things like the 
TeleYapper will give you a basis to work from.
 


Kerry 
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of TV 
  JOESent: Wednesday, April 26, 2006 7:31 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
  as a phone survey system
   Hi, I'm interested in developing an 
  automated phone survey and am curious  if Asterisk could be 
  configured to run such a system.. My idea is to  record a message and 
  a series of sub-questions. The system would  call each number on a 
  list and play the message, Depending on the touch tone response 
  another message would be played. Is it possible  for asterisk to 
  manage a survey like this? If so can the responses from  the 
  listeners be recorded. If someone else has done this I'd be 
  interested in details. TIA , TV JOE
  
  
  Yahoo! 
  Messenger with Voice. PC-to-Phone calls for ridiculously low 
rates.
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RE: [Asterisk-Users] Paging on Aastra analog phones.

[Kerry Garrison]

When used in this mode they can only detect a ring. Your 
best bet would be to put in some overhead paging.

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Richard 
  SchroederSent: Wednesday, April 26, 2006 6:51 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Paging on 
  Aastra analog phones.
  
  
  Hello 
  All,
   
  We have an installation that has 
  Aastra analog phones connected to the asterisk server with Sipura ATA devices. 
  (It was done this way in order to use existing 
  wiring).
   
  Is there any way to implement a 
  “page all” by turning on the speakers in these phones (like you can do by 
  turning on auto-answer on some SIP phones)?
   
  Thanks in advance for your 
  help.
   
  R C 
  Schroeder
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RE: [Asterisk-Users] USB conference phone

[Kerry Garrison]

it was a revewiers sample that I begged them to not make me 
send it back and they let me keep it.
 


Kerry 
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Wednesday, April 26, 2006 4:46 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] USB conference phone
  
  
  Lol – now the important 
  question….Did you pay for it or was it a reviewers sample J
   
   
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kerry GarrisonSent: Wednesday, 26 April 2006 7:23 
  PMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB 
  conference phone
   
  Yes, I have that 
  device, I wrote the review of it and have used it regularly ever since. I use 
  it with IDEFISK softphone for the most part but have tested it with Skype, 
  X-Lite, and SJPhone. I have had it since November and just love 
  it.
   
  Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
  
  (949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 
  
  
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Wednesday, April 26, 2006 8:24 
AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] USB 
conference phone
Kerry, do you actually own one? 
Have you used it for long? What are you using it 
for?
 
(jim – personally I cant see the 
point of using your phone when I have a very good quality headset and 
mic.).
 
 
Dean
 
 
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry GarrisonSent: Wednesday, 26 April 2006 10:36 
AMTo: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB 
conference phone
 
This is an 
excellent USB speakerphone
http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
 
 

   
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim HouserSent: Wednesday, April 26, 2006 6:26 
  AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB 
  conference phone
  I don't know 
  about this phone but I can tell you I have a vendor that will only talk to 
  me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm
   
  It 
  operates nice and has very good call 
  quality.
   
   
   
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 
  PMTo: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB 
  conference phone
  Has anyone actually used these 
  USB speakerphones
  http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
   
   
  Seems to get a pretty good 
  review here 
  http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
   
   
  But looking for real world 
  feedback.
   
   
  Cheers,
   
  Dean
   
  
  This e-mail and any attachments may contain 
  confidential and privileged information. If you are not the intended 
  recipient, please notify the sender, or [EMAIL PROTECTED], 
  immediately by return e-mail and destroy any copies. Any dissemination or 
  use of this information by a person other than the intended recipient is 
  unauthorized and may be illegal. Unless otherwise stated, opinions 
  expressed in this e-mail are those of the author and are not endorsed by 
  the author's employer. 
  
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RE: [Asterisk-Users] USB conference phone

[Kerry Garrison]

Yes, I have that device, I wrote the review of it and have 
used it regularly ever since. I use it with IDEFISK softphone for the most part 
but have tested it with Skype, X-Lite, and SJPhone. I have had it since November 
and just love it.
 


Kerry 
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Wednesday, April 26, 2006 8:24 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] USB conference phone
  
  
  Kerry, do you actually own one? 
  Have you used it for long? What are you using it 
  for?
   
  (jim – personally I cant see the 
  point of using your phone when I have a very good quality headset and 
  mic.).
   
   
  Dean
   
   
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kerry GarrisonSent: Wednesday, 26 April 2006 10:36 
  AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB 
  conference phone
   
  This is an excellent 
  USB speakerphone
  http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
   
   
  
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jim HouserSent: Wednesday, April 26, 2006 6:26 
AMTo: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB 
conference phone
I don't know about 
this phone but I can tell you I have a vendor that will only talk to me via 
Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm
 
It 
operates nice and has very good call 
quality.
 
 
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 
PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB 
conference phone
Has anyone actually used these 
USB speakerphones
http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
 
 
Seems to get a pretty good 
review here 
http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
 
 
But looking for real world 
feedback.
 
 
Cheers,
 
Dean
 

This e-mail and any attachments may contain 
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recipient, please notify the sender, or [EMAIL PROTECTED], 
immediately by return e-mail and destroy any copies. Any dissemination or 
use of this information by a person other than the intended recipient is 
unauthorized and may be illegal. Unless otherwise stated, opinions expressed 
in this e-mail are those of the author and are not endorsed by the author's 
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RE: [Asterisk-Users] USB conference phone

[Kerry Garrison]

This is an excellent USB speakerphone
http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
 
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim 
  HouserSent: Wednesday, April 26, 2006 6:26 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  RE: [Asterisk-Users] USB conference phone
  
  I don't know about this phone but I can tell you I have a 
  vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm
   
  It operates nice and has very good call 
  quality.
   
   
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dean 
  CollinsSent: Tuesday, April 25, 2006 8:22 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] USB conference phone
  
  
  Has anyone actually used these USB 
  speakerphones
  http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
   
   
  Seems to get a pretty good review 
  here 
  http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid=27
   
   
  But looking for real world 
  feedback.
   
   
  Cheers,
   
  Dean
   
  This e-mail and any attachments may contain confidential and privileged 
  information. If you are not the intended recipient, please notify the sender, 
  or [EMAIL PROTECTED], immediately by return e-mail and destroy 
  any copies. Any dissemination or use of this information by a person other 
  than the intended recipient is unauthorized and may be illegal. Unless 
  otherwise stated, opinions expressed in this e-mail are those of the author 
  and are not endorsed by the author's employer. 
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RE: [Asterisk-Users] Auto Logout from queue

[Kerry Garrison]

Yes, that is the functionality I am looking for, just not sure how exactly
to pull that off.


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Tuesday, April 25, 2006 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Auto Logout from queue


Use the local channel to call the agent first, and if there is no answer,
log them out.
 
 

  _  

From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Tue 4/25/2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Auto Logout from queue


i have a client that wants a function that will automatically logout an
agent from a queue if they do not answer a call. This would prevent future
calls from being sent to that phone if the agent forgot to logout. Any
ideas?
 
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 -  <mailto:[EMAIL PROTECTED]>
[EMAIL PROTECTED]
 <http://www.techdatapros.com/> http://www.techdatapros.com 
 

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[Asterisk-Users] Splitting Zap channels into trunks?

[Kerry Garrison]

On a TDM2400 with 3 
FXO modules, is there a way to split each line into basically being its own 
trunk or another way to pull off the following scenerio:
 
PBX has 12 inbound 
PSTN lines
1,3,5,7 are the 714 
phone number hunt group
2,4,6,8 are the 888 
phone number hunt group
9-12 are fax 
lines
 
Customer wants 
outbound calls to go out in the following order: 
8,7,6,5,4,3,2,1,12,10,11,9
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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[Asterisk-Users] Auto Logout from queue

[Kerry Garrison]

i have a client that 
wants a function that will automatically logout an agent from a queue if they do 
not answer a call. This would prevent future calls from being sent to that phone 
if the agent forgot to logout. Any ideas?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] About Softphone IAX free for Pocket PC

[Kerry Garrison]

Unless you have a top of the line Pocket PC don't even bother. Most
inexpensive units like the T-Mobile MDA just don’t have the processing power
to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I
forgot about already and the sound quality was horrible regardless of using
GPRS or WiFi. That would have been a great benefit to me but its just not
going to happen on a device that barely runs Windows Mobile as it is. 

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of makevuy
> Sent: Tuesday, April 25, 2006 8:03 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] About Softphone IAX free for Pocket PC
> 
> Hello,
> 
> Has anyone Knowledge about softphone IAX for pocket PC totally free?
> 
> Tkanks for all.
> 
> --
> Sandra Salmerón Ntutumu<[EMAIL PROTECTED]>
> Tlf. Analog: +34 914888405 / Móvil: 653574298 Tlf. IP desde 
> FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundación 
> EHAS: Enlace Hispanoamericano de Salud - www.ehas.org 
> Telemedicina rural para zonas aisladas de países en desarrollo
> 
> 
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RE: [Asterisk-Users] Polycom MWI

[Kerry Garrison]

Thanks a ton!!

When using Extensions mode (the default) this would be:

[EMAIL PROTECTED]

When Using Users and Devices mode this would be:

[EMAIL PROTECTED]

Thanks for the guidance there, this has been driving me nuts.

-Kerry

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Bill Gibbs
> Sent: Friday, April 21, 2006 5:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Polycom MWI
> 
> Ohh yeah good point.  I had a similar issue when I started 
> using FreePBX and it didn't fill out the mailbox field 
> automatically.  Once I added the [EMAIL PROTECTED] there the MWI 
> started working as well.
> 
> Bill
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Avi Miller
> Sent: Friday, April 21, 2006 12:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom MWI
> 
> Kerry Garrison wrote:
> > Didn't help. Could I be missing something else?
> 
> My phone.cfg looks like this:
> 
>msg.mwi.1.subscribe="300"
> msg.mwi.1.callBackMode="contact"
> msg.mwi.1.callBack="*97"/>
> 
> And sip.conf for extension 300:
> 
> [300]
> username=300
> type=friend
> secret=***
> record_out=Adhoc
> record_in=Adhoc
> qualify=no
> port=5060
> pickupgroup=1
> nat=never
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> disallow=all
> context=from-internal
> canreinvite=no
> callgroup=1
> callerid=Polycom IP501 <300>
> allow=alaw
> allow=g729
> 
> 
> Mine works fine, so I hope that helps. :)
> 
> --
> National Manager - Special Projects
> 
> < Sydney / Melbourne / Canberra / Hobart / London />
>2/340 Gore Street  T: +61 (0) 3 9486 0411
>Fitzroy, VIC   F: +61 (0) 3 9486 0611
>3065   W: http://www.squiz.net/
> 
> .>> Open Source  - Own it  -  Squiz.net ./> 
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RE: [Asterisk-Users] Polycom MWI

[Kerry Garrison]

Didn't help. Could I be missing something else?


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Thursday, April 20, 2006 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom MWI


Put your voicemailbox number (usually extension) in the 1.subscribe field.
 
Bill

  _  

From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 4/20/2006 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Polycom MWI


I have tried everything from voip-info and I still cant get the Polycom
501/601 to display the MWI indicator light. Everything else works just fine.
I am using FreePBX set to users and devices mode. Here is the MWI section of
the phonexxx.cfg file:
 
 
 
   
 
i have also tried
 
msg.mwi.1.callBackMode="register"
 
 
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 -  <mailto:[EMAIL PROTECTED]>
[EMAIL PROTECTED]
 <http://www.techdatapros.com/> http://www.techdatapros.com 
 

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[Asterisk-Users] Polycom MWI

[Kerry Garrison]

I have tried 
everything from voip-info and I still cant get the Polycom 501/601 to display 
the MWI indicator light. Everything else works just fine. I am using FreePBX set 
to users and devices mode. Here is the MWI section of the phonexxx.cfg 
file:
 

 
msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" 
msg.mwi.1.callBack="*97" msg.mwi.2.subscribe="" 
msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" 
msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="disabled" 
msg.mwi.3.callBack="" msg.mwi.4.subscribe="" 
msg.mwi.4.callBackMode="disabled" msg.mwi.4.callBack="" 
msg.mwi.5.subscribe="" msg.mwi.5.callBackMode="disabled" 
msg.mwi.5.callBack="" msg.mwi.6.subscribe="" 
msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/> 

 
   

 
i have also 
tried
 
msg.mwi.1.callBackMode="register"
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Announcement System for a Charity

[Kerry Garrison]

NerdVittles.com has a dialout announcement system article.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Nabeel Jafferali
> Sent: Thursday, April 20, 2006 11:40 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Announcement System for a Charity
> 
> I'm putting together an Asterisk server for a local charity 
> to use as an announcement system. I've been thinking about 
> how to write the dialplan to allow different options for 
> different groups' announcements, as well as mailboxes for the 
> various groups and the charity's administrators. Of course, 
> this would also need to include an option for the heads of 
> the different groups to modify their announcements.
> 
> Before I write it, I was wondering if anyone had an extensive 
> dialplan or an AGI script that already did something like 
> this. I know it'll only take a couple of hours to write and 
> test this, but I thought if someone has something already 
> written, I could just "borrow" it from you.
> 
> Thanks,
> 
> Nabeel
> 
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RE: [Asterisk-Users] need stand-alone FXO ports

[Kerry Garrison]

Linksys SPA-3000 Single Port $90
Mediatrix 1204 4 port Gateway $580
Rhino CB24 24 Port Channel Bank + Rhino R1T1 Card $2000

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Ronald Wiplinger
> Sent: Wednesday, April 19, 2006 12:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] need stand-alone FXO ports
> 
> What are you using as FXO ports for a few analog (remote) lines?
> What is the price, where to buy, what is your experience?
> 
> bye
> 
> Ronald Wiplinger
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RE: [Asterisk-Users] Asterisk no sound from sound card

[Kerry Garrison]

I don't know that distro but with CentOS you have to run alsamixer to turn
on the output and turn up the volume, it is off by default.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Gary Hodder
> Sent: Thursday, April 13, 2006 9:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Asterisk no sound from sound card
> 
> Hi all,
> 
> I have a k6 2-400 that has a creative awe64 sound card.
> It plays sound fine when using a standard audio player.
> I want to use the sound card as a console phone but there is 
> no audio from the sound card. I have tried compiling various 
> version of asterisk with no luck. I did use this machine 
> about 2 years ago with Asterisk and Mdk 10 and it worked 
> fine. Currently using Mandriva 2006.
> Any ideas in getting the sound card to work would be apreciated.
> 
> Thanks
> Gary.
> 
> 
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RE: [Asterisk-Users] freepbx dialing prefix

[Kerry Garrison]

Submit a bug report to the FreePBX 
team?

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sean 
  GarlandSent: Wednesday, April 12, 2006 8:46 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] freepbx 
  dialing prefix
  
  
  I need to put a ‘w’ 
  in the dialing prefix, but it says it isn’t valid.  If I manually modify 
  the extension file, it then affects all calls made over any trunk.  Any 
  ideas?
   
  Sean
  --No virus found in this outgoing message.Checked by 
  AVG Free Edition.Version: 7.1.385 / Virus Database: 268.4.1/309 - Release 
  Date: 4/11/2006
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[Asterisk-Users] SPA-941/942 Bulk provisioning

[Kerry Garrison]

Has anyone got any 
information on bulk provisioning of Linksys SPA-941/94s? There is an overview in 
the admin guide but it refers to a different provisioning guide that I haven't 
found anywhere.
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Instant Message?

[Kerry Garrison]

I tried the latest version of Jive over the weekend and I have to say it is
a giant pile of crap. I did this on multiple machines on both Linux and
Windows, and after setting everything up, the moment you add the asterisk
module, all authentication and user setup is lost and there is no way to log
back in as the admin to fix it. If anyone has any more positive experience I
would like to hear about it as it sounds very interesting.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steve Totaro
> Sent: Sunday, April 09, 2006 6:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Instant Message?
> 
> Zhiqiang Li wrote:
> > Hi all,
> >  
> > My client softphone supports IM feature. Does any warmheated expert 
> > know if Asterisk can support IM also at server side? If so, 
> is there 
> > any related documents or weblinks?
> >
> > --
> > Thanks & Best Regards!
> >
> > Steven Li
> >
> >   
> 
> I am not sure exactly what you are trying to do but Jive 
> Messenger has asterisk add-ons and functionality.  Might be 
> worth a look for ya. 
> 
> Thanks,
> Steve Totaro
> 
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RE: [Asterisk-Users] Force codec

[Kerry Garrison]

Disallow=all
allow=ulaw
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
  StrelnikovSent: Saturday, April 08, 2006 7:25 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Force 
  codec
  Hi,   Is it possible to force using codec depends 
  on extension? For example, voice codec is ILBC and with some prefix fax code 
  should be ulaw.Thanks.Best 
  regards,Michael 
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RE: [Asterisk-Users] increasing volume level to console/dsp

[Kerry Garrison]

Use an amplifier off the headphone jack. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jerry Geis
> Sent: Thursday, April 06, 2006 8:30 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] increasing volume level to console/dsp
> 
> I am trying to get higher volume on a console/dsp port.
> 
> I have a SIP phone connected to server A, Server A has an 
> IAX2 connection to Box B that connects me to the console/dsp 
> port. The mixer settings are set to 90% but the audio out is 
> not that high.
> 
> How can I increase the sound level.
> 
> Jerry
> 
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RE: [Asterisk-Users] Asterisk behind NAT

[Kerry Garrison]

Yes.

In Sip.conf you need the following lines:

externip=xxx.xxx.xxx.xxx ; put public ip address here
localnet=192.168.10.0/255.255.255.0 ; edit as appropriate

In your firewall, add the following mappings to your server:

5060-5061 UDP
10,000 - 20,000 UDP

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Joao Pereira
> Sent: Thursday, April 06, 2006 8:05 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Asterisk behind NAT
> 
> Hello to all
> Can we put Asterisk in a company that has an ADSL connection 
> with just one public IP address? Because with just one public 
> IP, Asterisk must have a private (NATed) IP... but the idea 
> is to make him dial other SIP domains.
> 
> Can Asterisk work behing NAT, and still route calls to the Internet?
> And he can still receive calls from the Internet?
> 
> Thanks
> Joao Pereira
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RE: [Asterisk-Users] Using Call Progress

[Kerry Garrison]

Welcome to the painful world of analog phone lines. Unless 
you are using a digital line, there really is no true call progress detection 
available. In many situations this is not a problem, where we see this the most 
is when you are trying to ring a zip device and a zap channel at the same time, 
the zap call progress indicates an answered line the moment the zap channel goes 
active, NOT when the far side answers. If you have a ring group with sip and zap 
channels, what typically happens is that the sip phone will ring once, but as 
soon as the TDM card places the outbound call, it is considered "answered" and 
the sip phone stops ringing. Yes, you can enable callprogress and several other 
tweaks but the end result is often the far side answering and Asterisk still 
playing ring tones because there is no signal on the PSTN to indicate a far side 
answer.
 
So, what to do when you find yourself in this situation and 
adding a PRI is not a solution, the only way we have worked around this is to 
make those outbound calls over a SIP or IAX service provider (and no, using a 
SIP gateway like a Mediatrix 1204 does not solve the problem as it is a PSTN 
issue)
 
I know some people will argue this, but this was the result 
of almost 12 hours of work with us and Digium to figure out this issue. After 
MUCH debate and many hours of testing, this became the official 
word.
 
Don't shoot the messenger.
 
Kerry 
GarrisonDirector of Technical ServicesTech Data 
Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Eric 
  BuruschkinSent: Thursday, April 06, 2006 6:19 AMTo: 
  Asterisk-UsersSubject: [Asterisk-Users] Using Call 
  Progress
  
  
  I'm attempting to use callprogress in my system, 
  and I'm having trouble.   Callprogress always can tell if the line 
  is busy or ringing, but when the line is answered, the call does not get 
  bridged.  Messages showing that "line is ringing" stop in the console and 
  if the called party hangs up, asterisk reports the line is busy.
   
  Are there any settings that I could use to help 
  with this issue?  I am using asterisk 1.2.4 with TDM04B (FXO) cards on a 
  RHEL3 system.  Something in indications.conf or zonedata.c/dsp.c in the 
  source that can be tweaked?
   
  Any help would be appreciated!
   
  Thanks!
   
  - Eric Buruschkin
   
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RE: [Asterisk-Users] Beginner: PBX for my house

[Kerry Garrison]

"or you can use hardware devices to connect to 
traditional phone lines"
 
That can be a Digium card, Sangoma card, Linsys 
SPA3000, Mediatrix 1204, and several other devices.
 
Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  RandyWSent: Monday, April 03, 2006 8:31 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionCc: 
  'erik'Subject: Re: [Asterisk-Users] Beginner: PBX for my 
  house
  What about the Asterisk Developer pack, a good Linux box and his 
  standard phone line?? I've seen this work and it does a great job.  The 
  Telco doesn't know anything as Asterisk integrates with the analog phone line 
  and things just work.Am I off base here??RandyWKerry 
  Garrison wrote: 
  



Currently Asterisk will not integrate with Skype. You 
would need a provider such as Teliax, Broadvoice, IAX.cc or many others or 
you can use hardware devices to connect to traditional phone lines. You 
didn't say what broadband phone you have but if its Vonage, there are also 
issues with "true" integration there as well.
 
There are lots of good articles at VOIPSpeak.net and 
NerdVittles.com
 
Kerry 
GarrisonDirector of Technical ServicesTech 
Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - 
[EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of erikSent: Monday, April 03, 2006 7:40 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Beginner: PBX for my house
  Hi,
   
  I am renovating my house completely and 
  installing new cabling for communication. 
   
  I'm not to into this PBX thing but I would 
  like to have a simple one for my house, to have different phone numbers 
  for my family members, some kind of integration with my Broadband 
  telephone, and possibly Skype  
   
  Browsing the digium website didn't 
  make me much wiser. I have a linux box that I can dedicate to this. If I 
  buy for example the Wildcard TE110P card, what can I as a simple homeuser 
  do with this card?  I have reasonable knowledge about networking and 
  Linux administration, but next to little about digital telephony 
  and such.
   
  erik
   
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RE: [Asterisk-Users] Beginner: PBX for my house

[Kerry Garrison]

Currently Asterisk will not integrate with Skype. You would 
need a provider such as Teliax, Broadvoice, IAX.cc or many others or you can use 
hardware devices to connect to traditional phone lines. You didn't say what 
broadband phone you have but if its Vonage, there are also issues with "true" 
integration there as well.
 
There are lots of good articles at VOIPSpeak.net and 
NerdVittles.com
 
Kerry 
GarrisonDirector of Technical ServicesTech Data 
Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  erikSent: Monday, April 03, 2006 7:40 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Beginner: 
  PBX for my house
  
  Hi,
   
  I am renovating my house completely and 
  installing new cabling for communication. 
   
  I'm not to into this PBX thing but I would like 
  to have a simple one for my house, to have different phone numbers for my 
  family members, some kind of integration with my Broadband telephone, and 
  possibly Skype  
   
  Browsing the digium website didn't make 
  me much wiser. I have a linux box that I can dedicate to this. If I buy for 
  example the Wildcard TE110P card, what can I as a simple homeuser do with this 
  card?  I have reasonable knowledge about networking and Linux 
  administration, but next to little about digital telephony and 
  such.
   
  erik
   
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RE: [Asterisk-Users] Asterisk 2.0 Where to download

[Kerry Garrison]

Think people will fall for it again next year too? 

> > Hello All
> >
> >
> >   I read in www.sineapps.com have Asterisk 2.0 rewritten C# 
> and run on 
> > windows, any body could be mail or send to me URL to download.
> >
> >   Thanks
> > 
> > 
> > Tin Trung Nguyen
> > Technical Team
> > Mobile: 084-91.365.4857
> > website: www.daivietcontrol.net>
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> >   
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> __
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> around http://mail.yahoo.com 
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RE: [Asterisk-Users] How is Teliax ?

[Kerry Garrison]

When it works it works great. We have had a few issues 
lately but they were resolved fairly quickly.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
  Reddy BandiSent: Thursday, March 30, 2006 8:14 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] How is 
  Teliax ?
  
  Hi I am looking at purchasing some DID 
  lines from Teliax to install it on my asterisk.i would like to know some 
  feed back on "Teliax" before i purchase.suggest me if there are better 
  sevice providers.thanksGiridhar Bandi 

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RE: [Asterisk-Users] Marketing Materials

[Kerry Garrison]

http://www.asterisk.org/features 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Wednesday, March 29, 2006 3:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Marketing Materials
> 
> Darrick Hartman wrote:
> > Bob McDowell wrote:
> >> The owner of my company just asked me for an Asterisk 
> brochure.  Has 
> >> anyone seen such a creature?  I know of some really informative 
> >> websites, but I think a pdf would be priceless at this point.
> >>
> > 
> > Bob,
> > 
> > Check on Digium's website.  I know there is such a creature there.
> > 
> > Darrick
> 
> Just went looking and could not find a thing. Can you give us a url?
> 
> 
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RE: [Asterisk-Users] Dumb question - reaching the PSTN

[Kerry Garrison]

With Asterisk you can use Analog lines (PSTN) , Digital lines (PRI), or
Internet Telephone Service Providers (ITSP) such as Broadvoice, Teliax,
IAX.cc, and many more.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Charles Marcus
> Sent: Wednesday, March 29, 2006 2:38 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Dumb question - reaching the PSTN
> 
> Hi everyone,
> 
> I am fairly new to the idea of VoIP, although I've been 
> reading about it off and on for the last few years. Now it is 
> starting to look mature enough to consider implementing it, 
> but there is one thing that I haven't been able to get a 
> clear answer on...
> 
> With Vonage, you are using the Vonage network - it is their 
> responsibility to route your call to the endpoint, which is 
> more than likely on the old fashined PSTN.
> 
> If I install Asterisk, how do my calls actually get 
> completed? How do they get 'bridged' over to the PSTN?
> 
> I attended a Seminar today hosted by Dynasis, and one of the 
> issues was VoIP. ShoreTel was there, and the said I had to 
> have phone lines, whether they were POTS lines, chennels from 
> a T-1, whatever, we still had to have phone lines.
> 
> Now I'm confused.
> 
> If I implement an Asterisk based system (yes, I'd be paying a 
> consultant to help), will I still have to maintain phone 
> lines and pay full price for Long Distance?
> 
> Simple pointers to White Papers on this issue will be sufficient.
> 
> Many thanks,
> 
> -- 
> 
> Best regards,
> 
> Charles
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RE: [Asterisk-Users] AAH lost my IVR phrases

[Kerry Garrison]

You made some change to something using AMP and it overwrote the
extensions_additional.conf file as it was designed to do. The only safe
place to put customizations is in extensions_custom.conf.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jim Hanlon
> Sent: Wednesday, March 29, 2006 8:02 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] AAH lost my IVR phrases
> 
> Hello-
>  I have a low traffic AAH setup, a few hardphones, a few 
> softphones, 50 calls per day max. I used the AMP Digital 
> Receptionist to make a simple voice menu: "Thank you for 
> calling ". I did this for both Normal times and After 
> Hours times. It worked fine.
> 
> I then went to the AMP Maintenance window, Config Edit, got 
> the "phpconfig for Asterisk PBX" page, and selected the 
> extensions_additional.conf page. On this page were the 
> entries for the Normal and After Hours greetings. The initial 
> greeting phrases were expressed in terms of statements like:
>  "exten => s,n,Background(custom/aa_num)". It was easy to 
> extend the greeting (for instance, "Office hours are 7-7", 
> "Press pound for directory"..) by directly adding more canned 
> phrases, like so:
>  "exten => s, n+1, Background(custom/aa_num+1)"
>  etc...
> 
> Hit update, Re-Read Configs. Try it out. 
> 
> It worked fine. And I felt pretty clever. For a few weeks.
> 
> Then a complaint: Callers encountered an obviously truncated 
> IVR script, and had no way out of the maze. Sure enough, only 
> one phrase was being uttered. And, sure enough, only one 
> phrase was being commanded by the existing 
> extensions_additional.conf file. I re-edited the file, 
> updated, and things worked again.
> 
> !!!? What happened to my edited, updated, and Re-Read 
> extensions_additional.conf file?
> 
> Anybody ever encounter this behavior?
> 
> What to do, in order to avoid this mishap in the future?
> 
> Ideas, thoughts?
> 
> Thanks,
> 
> Jim Hanlon
> 
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[Asterisk-Users] Dell 2850 w/TDM2400?

[Kerry Garrison]

Does anyone know if 
a TDM2400 will fit into a Dell 2850? 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] FreePBX & AAH

[Kerry Garrison]

FreePBX is a configuration manager for Asterisk. It is NOT its own version
of Asterisk, it is simply a GUI to manage the config files.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Waldo Rubinstein
> Sent: Monday, March 27, 2006 9:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] FreePBX & AAH
> 
> Pardon the question, but what I understand of FreePBX is that 
> it's basically Asterisk with a web interface and some 
> additional modules.  
> Is that correct? Can you install FreePBX on a system which 
> ALREADY has asterisk up and running or does it require ITS 
> version of asterisk?
> 
> Thanks,
> Waldo
> 
> On Mar 27, 2006, at 12:29 PM, Tom Vile wrote:
> 
> > Yes, you can.
> >
> > On 3/27/06, Jim Houser <[EMAIL PROTECTED]> wrote:
> >> Does anyone know if FreePBX can be installed on a Linux 
> box that was 
> >> built using [EMAIL PROTECTED]  I would prefer to manage Asterisk with 
> >> FreePBX over
> >> the AAH build.   I have just not had good luck building an  
> >> Asterisk system
> >> from scratch and the Centos based Amp ISO and prebuilt 
> config files 
> >> are a wonderful place to start.  Nothing against Asterisk 
> or Linux.  
> >> My build from scratch issues are only due to my lack of Linux 
> >> experience...
> >>
> >> Thanks
> >>
> >>
> >>
> >> This e-mail and any attachments may contain confidential and 
> >> privileged information.  If you are not the intended recipient, 
> >> please notify the sender, or [EMAIL PROTECTED], 
> >> immediately by return e-mail and destroy any copies. Any 
> >> dissemination or use of this information by a person other 
> than the 
> >> intended recipient is unauthorized and may be illegal.  Unless 
> >> otherwise stated, opinions expressed in this e-mail are 
> those of the 
> >> author and are not endorsed by the author's employer.
> >>
> >>
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> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > --
> > Tom Vile
> > Baldwin Technology Solutions, Inc
> > Consulting - Web Design - VoIP Telephony 
> www.baldwintechsolutions.com
> > Phone: 518-631-2855 x205
> > Fax: 518-631-2856
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RE: [Asterisk-Users] Re: Best GUI for basic HostedPBX service

[Kerry Garrison]

FreePBX allows you to set up multiple companies as well as determine what
level of access each user has.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tomislav Parcina
> Sent: Sunday, March 26, 2006 10:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Re: Best GUI for basic HostedPBX service
> 
> In article <[EMAIL PROTECTED]>, 
> [EMAIL PROTECTED] says...
> > Hi,
> > 
> > I'm looking for a web GUI to offer my end-users (Hosted PBX), and I 
> > thought I'd pick a few brains first.
> > 
> > I'm not looking to configure the Asterisk server itself, VI works 
> > adequately for that.  But I want to give Web access to as 
> many of the 
> > following
> > features:
> 
> This is something I'm will need in few months. If you find 
> anything, please let the group know.
> 
> 
> --
> Tomislav Parcina
> tparcina#lama.hr
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RE: [Asterisk-Users] 3Com Phones

[Kerry Garrison]

Look at the Linksys SPA942, it's a great phone for the price. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Radcliffe
> Sent: Sunday, March 26, 2006 10:21 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] 3Com Phones
> 
>Hi Daniel,
> 
>If you are not locked in to an asterisk solution, I have a 
> friend I have done a couple of network/phone systems with.  I 
> am also looking at Asterisk but have not gotten into it that far.
> 
>Rich Radcliffe
>Kondor Waffenamt
>(760) 240-4728
>[EMAIL PROTECTED]
>  
> 
> >>> [EMAIL PROTECTED] 3/26/2006 9:55:38 AM >>>
> Drat, because the 3Com phones looked pretty good for the price. :)   
> Is there somewhere that has a compatibility list for Asterisk 
> with all the phones that are known to work/not work with 
> Asterisk; since apparently VoIP phone companies incorrectly 
> state that they support the SIP protocol (I don't consider, 
> "we support SIP as long as it only talks to our server 
> because we tweaked it just a bit" to be "supported").
> 
> I am looking for a good 60 phones.  We are upgrading our entire phone 
> 
> system (and *old* NEC PBX).  We don't need anything fancy on 
> most of the phones, just the "usual" mid-size business features.  
> Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 
> 
> attendant stations that can see all in-use phone lines.  We 
> are trying to keep the costs (relatively) down, hence using 
> Asterisk instead of a full commercial solution.  It is very 
> disconcerting to know the providers are essentially lying 
> about what their phones support. (3Com states their phones 
> are SIP compatible, not 3Com's version of SIP compatibile).
> 
> Thanks for the info, hopefully somebody will have some 
> recommendations for a good phone brand that actually IS 
> Asterisk compatible.
> 
> Daniel
> 
> On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote:
> 
> > I would not recommend the 3Com phones for use with Asterisk.
> >
> > 3Com 3100 series phones do not support SIP with non-3Com systems.   
> > They have
> > a basic boot loader which must download code from a 3Com 
> NBX or a 3Com 
> > VCX system.  If you don't have either of these, then you won't get 
> > runtime code on the phone, thereby making it impossible to use the 
> > thing with Asterisk.
> >
> > I've heard rumors that the 3103 phones have enough storage space on
> 
> > the
> > phone to store a SIP image, but I don't have any more 
> information than 
> > that.
> >
> >
> > As far as 3Com licensing is concerned, it's not per year, it's per- 
> > seat (one-time charge), just like any other commercial VoIP 
> PBX vendor 
> > (Cisco, Avaya, Shoretel, etc.)
> >
> > Jared Valentine
> > [EMAIL PROTECTED]
> 
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RE: [Asterisk-Users] simple question on asterisk

[Kerry Garrison]

Its all about how you configure your dialplan. Asterisk doesn't know what a
PSTN or VOIP phone number is. If you want all 08444 numbers to go through a
certain trunk, then you set your dialplan up accordingly.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Mark Hayward
> Sent: Monday, March 20, 2006 8:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] simple question on asterisk
> 
> Hi,
> I am planning to deploy an asterisk installation but I need 
> to convince a few managers that its a good idea.
> Theres something I don't quite understand though, I plan 
> deploy a box on the end of 4 channel BRI ISDN and provide it 
> an ADSL internet connection.
> Should a phone behind the asterisk PBX wish to call a VOIP 
> phone number number, say an 0844 one from www.voip-user.org, 
> would it send this automatically over the PSTN ISDN network 
> or would it know to send the call over the internet.
> Would I need a SIP provider on the internet to forward the 
> calls? I assume I would need some sort of directory service 
> to know where to route the call.
> Thanks in advance,
> Mark
> 
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[Asterisk-Users] Asterisk Users Group Tonight, Irvine, Ca

[Kerry Garrison]

If you are in Southern California and would like to attend the Asterisk Users 
Group Meeting, it is tonight from 6-9pm at the Heritage Park Library.
Irvine Heritage Park Library(949) 936-404014361 Yale AveIrvine, 
CA 92604
Tonight we will be having a demo of SIPX, a review of the SNOM 320 phone, and 
a look at FreePBX, the new version of the Asterisk Management Portal. Also, more 
books to give away from O'Rielly!!
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

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RE: [Asterisk-Users] Attended call transfer with GXP-2000

[Kerry Garrison]

If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNF
and hitting Line 1 will transfer Line 2 to Line 1. Same concept as
Conference. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
> Sent: Thursday, March 16, 2006 7:30 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Attended call transfer with GXP-2000
> 
> Can someone explain me attended transfer with Grandstream GXP-2000?
> Hitting TRNF button, I get:
>  Dial number (BLIND) or
>  Select line (ATTENDED)
> What's the exact meaning of 'Select line'?
> 
> Thanks
> Mimmus
> 
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[Asterisk-Users] Asterisk Users Group Meeting March 16, Irvine, Ca

[Kerry Garrison]

Irvine California, Heritage Park Library on the 
corner of Yale and Walnut. The Walnut is just south of the 5 fwy and Yale is 
between the Culver and Jeffery offramps. Meeting will run from 6 - 9pm. This 
week will feature a review of the SNOM 320, a demo of SIPX, some book giveaways 
courtesy of O'Rielly, and much more. 
 
For more information, contact me
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Latest Dell SC430 Compatibility With Wildcard

[Kerry Garrison]

If you go into the BIOS and disable all unneeded devices (serial, parallel,
USB, floppy, etc) then you shouldn't have any problem. I have one in a 15
user setup that is working fine.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Leo Ann Boon
> Sent: Tuesday, March 14, 2006 5:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Latest Dell SC430 Compatibility 
> With Wildcard
> 
> Anyone knows if the SC430, based on the Intel E7230 chipset, 
> is compatible with the Digium cards? I've tried the 
> compatibility page on digium's website. It seems like they've 
> pulled the old compatibility list, now the links on the page 
> only point back to the product pages. 
> Over here, Dell is selling (for a short period of time), 
> SC430 with Pentium D 820 Dual Core Processor 2.8GHz, 256MB 
> RAM, 80GB SATA for about US$240.
> 
> 
> 
> 
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RE: [Asterisk-Users] Analog Desktop Phone

[Kerry Garrison]

You really aren't going to find an analog phone that works as well as a SIP
phone for what you are trying to do. Some people suggested the GXP2000 for
$85 which works ok in a home environment. It is not a top quality phone but
it has all the features you want plus works very nicely with Asterisk. 

This same conversation is constantly going on on numerous forums. If you
think about what you are trying to accomplish, it might put things into
perspective. You are taking a state-of-the-art phone system flush with every
business feature you may ever want and trying to install it into your home
and you want to use a cheap phone on it. Things are just not designed that
way. If you want to be happy with your system, not to mention putting some
value on your time (and heaven help you if you have a wife that will use the
system) you do NOT want to use a cheap phone on this system. At a minimum go
with a Linksys SPA941 or a Snom 360. You will have either one working in a
matter of minutes. If you don't put any value on your time, then keep
monkeying around with a lesser solution, but the few hours you will save
just dropping in a decent phone should more than make up for the extra cost.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Thczv F. Thczv
> Sent: Monday, March 13, 2006 8:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Analog Desktop Phone
> 
> On 3/12/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
> 
> > > But what the OP wanted was a sulotion that together with 
> the SAP3000 
> > > makes for something that works even when there is a 
> blackout, since 
> > > the SPA3000 allows for failover to the FXS port from the FXO port 
> > > if/when there is no power to the unit. Which makes it a very good 
> > > solution when needed because of 911 reasons or the like.
> 
> > Actually it seems to me the Sipura 3000 is overkill in that case.
> > There are many other ATA's that are less expensive that 
> also have a 1 
> > port FXS, and a PSTN failover for blackout. It seems the OP doesn't 
> > need the FXO at all?
> >
> > The PA168V based ATA I have does this and was a little more 
> then half 
> > the cost of a SPA3000.  Works well too.
> 
> Shame on me, but I already have the SPA3000.  I like it very 
> much and it works fine.  Perhaps if I need another, I will 
> look at different products.
> 
> This is for my own home, where I am keeping my POTS line, partly as a
> 911 solution.  I have found a lot of analog desktop phones 
> that have some of the features I want, but not all of them.  
> The Cortelco 2200 looks like it might fit the bill.  But it 
> costs about $80.  I'm not sure I want to pay that much for an 
> analog phone that isn't wireless. 
> Other than that, the closest I have found so far is the AT&T 959:
> 
> http://www.amazon.com/gp/product/B00067KETY/ref=wl_it_dp/104-2
> 261851-2083919?%5Fencoding=UTF8&colid=1SGHZOJ18P2FB&coliid=I23
IRSR1SF2HPG&v=glance&n=172282
> 
> The problem with that one is that (as near as I can tell from 
> the photos and the manual) it has no visual MWI.
> 
> Still looking,
> 
> Dave
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[Asterisk-Users] Echo Cancelation on TE110P

[Kerry Garrison]

On a 55 station 
install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are 
complaiining about echo. According to the users, the echo seems to be phone 
number dependant. They claim that certain phone numbers have echo while others 
dont. Are there any tuning parametes like there is for a TDM400 card? 

 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Multiple TDM400P's in a single machine

[Kerry Garrison]

Yes you can but it is generally not a good idea nor is it simple to resolve
the additional IRQ conflicts on some machines.
-Kerry
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Thomas Artner
> Sent: Monday, February 20, 2006 4:38 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Multiple TDM400P's in a single machine
> 
> Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
> > Hi All,
> >
> >
> >
> > Can someone give me a definite answer as to wether or not you can 
> > reliably run multiple TDM400P's in the same machine?
> >
> > I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing 
> > key system, but I have seen several threads suggesting that this is 
> > not a supported configuration
> >
> >
> 
> i have two tdm400p's  (2xFXO, 6xFXS) in one desktop machine 
> used as asterisk server for a small office (so the pc 
> hardware is nothing special).
> This configuration is running since two weeks without any problems!
> 
> 
> 
> >
> > Thanks,
> >
> >
> >
> > Marc.
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[Asterisk-Users] SPA-941 stutter tone

[Kerry Garrison]

I dont recall the 
SPA-941 playing a stutter tone in the previous firmware but it is driving me 
nuts, anyone know where to turn it off?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Solution for 1 time blast of 200, 000 recordedcalls

[Kerry Garrison]

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Tuesday, February 14, 2006 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Solution for 1 time blast of 
> 200, 000 recordedcalls
> 

SineDialer, ViciDial, and GNUDialer can all do this.
-Kerry


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[Asterisk-Users] AAH 2.5 pone paging broken

[Kerry Garrison]

Using some scripts 
that have been posted, we have been able to get paging to phones working quite 
nicely. However, with a few [EMAIL PROTECTED] 2.5 installs, (Aserisk 1.2.4) the 
phones ring but never pick up. Any ideas on why or how to tweak the scripts to 
get the phone paging working again?
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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RE: [Asterisk-Users] Asterisk vs. Traditional PBX

[Kerry Garrison]

Title: Asterisk vs. Traditional PBX



Hi everyone !So here's my question 
of the day !  I need to make a decision on whether or not to go to a voip 
solution or configure an existing pbx (norstar) that my company has 
available.  We are a small startup. I'm wanting a solution that will 
support up to about 200 people, with direct dial-in capability, up to about 30 
concurrent phone calls and good voice quality. Right now I have an asterisk 
deployment with about 15 people on it. We have sipura 841 phones. The biggest 
issue currently is voice quality. lot of complaints there.  I have a dell 
650 poweredge (single processory system), with a digium tdm400 card and 4 analog 
lines plugged into it.
[Kerry Garrison 
sayeth] 
The 841 is fine for testing but I would never put 
one on a clients desk. The sound quality is bottom of the barrel. Combine that 
with the TDM400 card and its a wonder anyone will use the phone system at 
all. Move up to the Linksys SPA941 or SPA942 or the Polycom 501 and then 
use a different interface such as the Mediatrix 1204 or a PRI and your 
users will be singing your praises till the end of 
time. So here are my questions:* Is asterisk 
a good solution for my company ? or should I just install the traditional pbx 
and look to move to asterisk in a couple of years ? (I personally would prefer 
asterisk cuz I'm a  unix person not a phone person so from a manageability 
perspective i would love this )[Kerry Garrison 
sayeth] 
Asterisk is a great solution for your company and you 
will have many more benefits than the Northstar system. 
 * If I were to go to an asterisk 
solution to support about 200 people with the requirements above what hardware 
platform would you recommend ?  I'm guessing I'd need a PRI line and a 
different digium card? Also would a 1cpu poweredge dell be enough ? or would 
that have to be upgraded too ? 
[Kerry Garrison 
sayeth] 
You would want a beefier machine and at least one PRI. 
Its not the number of people, its the number of concurrent phone calls. I see 
businesses with 100 people and they average 5-7 concurrent calls and I have 
clients with 15 people that average 12-15 concurrent calls. 
 If anyone is running an environment similar to this 
that can provide help I would really appreciate this. I'm having a hard time 
making this decision and would love to hear anybody's experience in a real time 
environment.
[Kerry Garrison 
sayeth] 
My largest install is approaching 55 users, with the 
PRI and Polycom 501's they couldnt be happier. The system is on a nice 2.8ghz 
XEON system with 2gb of RAM and at peak times the server is basically 
idle. Thanks again this list ROCKS!Nora 
Lavelle
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RE: [Asterisk-Users] Handset phone to replace Flash Operator Panel

[Kerry Garrison]

The best way I have found to use FOP is on a second monitor. That way you
don't need a second PC and it doesn't run behind the receptionists other
applications. Just a decent LCD monitor and a second video card or a dual
head video card, and you are all set.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Garth van Sittert
> Sent: Tuesday, February 07, 2006 9:46 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Handset phone to replace Flash 
> Operator Panel
> 
> Hi All
> 
> Has anyone come across a handset that can somehow replace 
> FOP?  Some users don't like FOP unless it is on a dedicated PC.
> 
> Thanks
> Garth
> 
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RE: [Asterisk-Users] virtual extension per user ?

[Kerry Garrison]

This can easily be accomplished with AMP using the Users and Devices mode. 
http://voipspeak.net/index.php?/content/view/49/28/
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Alex Ongena
> Sent: Tuesday, February 07, 2006 8:55 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] virtual extension per user ?
> 
> certainly on his first call, but it should be possible for 
> him to explicitly 'register' and 'unregister'
> 
> On Tuesday 07 February 2006 17:06, Joe Tahan wrote:
> > when exactly would you like to stream this "register me" thingy? 
> > whenever an employee picks up the phone to dial? or when? 
> Please specify more.
> >
> > Truely/
> > Joe
> >
> >  From: Alex Ongena <[EMAIL PROTECTED]>
> > Reply-To: Asterisk Users Mailing List - Non-Commercial 
> > Discussion To: Asterisk 
> > 
> > Subject: [Asterisk-Users] virtual extension per user ?
> > Date: Tue, 7 Feb 2006 15:26:23 +0100
> >
> > >Hi,
> > >
> > >People here often work on 2-3 places (office 1, office 2 and home).
> > >
> > >I would like to give them 1 extension (XXX) and to ask them to 
> > >'register' the phone they use at a certain moment.
> > >
> > >The idea is that, when you need someone, just dial XXX and 
> the phone 
> > >near him (in Office 1, Office 2 or at Home), will ring.
> > >This will keep my queue system and other tricks intact, where I 
> > >always use the single extension XXX.
> > >
> > >I know you can 'forward' calls to other extensions, but 
> when people 
> > >go from Office 1 to Office 2, they forget to enable their 
> forward in 
> > >Office 1 to Office 2.
> > >I like a solution where they can say 'Please register me, I'am now 
> > >sitting in Office 2'. The moment after 'registration', 
> when you call 
> > >XXX, the phone in Office 2 will ring.
> > >
> > >In all places I use Asterisk 1.2.1 with bristuff, Cisco 7940/60 
> > >phones with Sip and some Sip softphones.
> > >
> > >Any hints or tricks to get this behaviour ?
> > >
> > >Thanks
> > >Alex
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> >
> > Don't just Search. Find! Try MSN Search:  Fast. Clear. Easy.
> 
> --
> Alex Ongena
> Managing Director
> ---
> Able N.V.Tel: +32(0)15 50.44.00
> Dellingstraat 28bFax: +32(0)15.50.44.09
> B-2800 Mechelen
> Belgium  mailto:[EMAIL PROTECTED]
> http://www.axsguard.com http://www.doITsafe.net
> 
> aXs GUARD - internet communication appliance
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