Re: [asterisk-users] How do I find a phone numbers issued by Rogers?

[Robert Augustyn]

Hm,
I did not think about that  I just assumed that they would not give it as 
this are the contact number of their clients ...
I believe that I have seen it somewhere on the web cannot find it though.

Sincerely,

Robert Augustyn
p:519.997.3106 ext:802
m:519.817.2503
e:robert.augus...@linqone.com



-Original Message-
From: shma...@gmail.com [mailto:asterisk-users-boun...@lists.digium.com] On 
Behalf Of C F
Sent: Wednesday, March 02, 2011 11:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I find a phone numbers issued by Rogers?

Call them.

On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn
 wrote:
> Hi,
>
> Is there a way of finding out what block of phone numbers were issued to
> Roger?s business customers in my end of the woods?
>
> Thanks,
>
> Sincerely,
>
>
>
> Robert Augustyn
>
> --
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[asterisk-users] DID's for Chatham, ON

[Robert Augustyn]

Can anybody provide DIDs for Chatham, ON?

Usage based preferred, but flat-rate is not an issue.

 

 

Contact off list.

 

Thanks for your time,

 

 

Sincerely,

Robert Augustyn

 

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[asterisk-users] How do I take out one office out of the call stream?

[Robert Augustyn]

Hi, 

I have two locations A and B. 

I have calls coming in into both locations but they are answered only by 
location A, location B forwards all calls to location A to be answered. 

Now when I have a call coming into location B then the call gets transferred to 
Location A then transferred to location B again it seem like the location A is 
still in the stream. 

Is there a way of taking it out of the stream? 

Thanks, 

robert 

  

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Re: [asterisk-users] What is the best way to configure this?

[Robert Augustyn]

I believe that for this, box B would have to answer the pstn line first …. I do 
not want that to happen 

  

Another question I have is how to configure Aastra phones to work with both 
servers and continue to work when internet connection is down? 

Thanks 

  

  

From:da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On 
Behalf Of Danny Nicholas
Sent: Monday, October 26, 2009 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] What is the best way to configure this? 



  

One suggestion – use “ex-girlfriend” logic on server b to only allow pickup of 
calls from Server A. 

  

-   exten => s,1/5551212,Answer 

-   exten => s,n,Hangup 

  

  


From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn
Sent: Monday, October 26, 2009 3:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What is the best way to configure this?


  

Hi, 

I have two servers ( A and B) in different towns. 

Both servers have pstn attached to them. Now I need to have calls coming to 
both servers to be answered on server A and then distributed between two sites. 

What is the best way of doing ? 

Having all calls to B forwarded to A on telco’s end? Having calls from B 
forwarded to A through internet? 

  

I tried to call forward the line on telco’s end but it still comes to the 
asterisk box and asterisk answers it generating clicking noise which I would 
like to avoid. 

How do I stop asterisk from answering that line? 

  

Thanks 

  

  

  

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[asterisk-users] What is the best way to configure this?

[Robert Augustyn]

Hi, 

I have two servers ( A and B) in different towns. 

Both servers have pstn attached to them. Now I need to have calls coming to 
both servers to be answered on server A and then distributed between two sites. 

What is the best way of doing ? 

Having all calls to B forwarded to A on telco’s end? Having calls from B 
forwarded to A through internet? 

  

I tried to call forward the line on telco’s end but it still comes to the 
asterisk box and asterisk answers it generating clicking noise which I would 
like to avoid. 

How do I stop asterisk from answering that line? 

  

Thanks 

  

  

  

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Re: [asterisk-users] I am looking for a good source of Monterrey DIDs

[Robert Augustyn]

good point :) 
 




From: abalas...@evaristesys.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Wednesday, April 29, 2009 1:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] I am looking for a good source of Monterrey DIDs



I cordially point you to asterisk-biz.

--
Sent from mobile device

On Apr 28, 2009, at 9:39 PM, Robert Augustyn < robert.augus...@linqone.com > 
wrote:



Any pointers will be appreciated... 
 
Sincerely, 
Robert Augustyn 






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[asterisk-users] I am looking for a good source of Monterrey DIDs

[Robert Augustyn]

Any pointers will be appreciated... 
 
Sincerely, 
Robert Augustyn 




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Re: [asterisk-users] How to verify availability of the DID connection?

[Robert Augustyn]

Thanks,
Well sometimes I have a situation that the trunk is registered but there is no 
communication coming in.
So ping and looking for registration status does not work ...
When I run sip reload it starts working again ?
One difference is that I can see is the refresh on the registration is 585 and 
not the usual 105.
Can I adjust this down anywhere?


Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: supp...@ocg.ca [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of OCG Technical Support
Sent: Saturday, March 07, 2009 9:59 AM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] How to verify availability of the DID connection?

Robert,

We've helped clients setup monitoring scripts for this type of situation - 2 
different ways.  One is a ping script, the other monitors the asterisk peer 
status of registration.  These were temporary until they could get to the root 
cause however.  Since you have multiple providers going down, I would dig into 
the cause on your end...

What diagnostics have you done so far?

Michelle Dupuis
www.generationd.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn
Sent: March 7, 2009 9:36 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] How to verify availability of the DID connection?

All these questions are valid, though I want first to see that the DID does not 
work then I will go and try to resolve it.  
I do not have a specific issue at this moment.

Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: asterisk@sedwards.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, March 06, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to verify availability of the DID connection?

On Thu, 5 Mar 2009, Robert Augustyn wrote:

> Occasionally, DIDs from different providers stop working for some 
> reason.
>
> I would like to be able to monitor situations like that and react 
> before any of my clients start going ballistic on me.

Are you losing DIDs that terminate on your Asterisk box or your clients 
Asterisk box?

Are these DIDs registering with Asterisk and are you re-registering often 
enough?

Is it a problem within the providers? Can you port the DIDs to another provider?

Why do the DIDs stop working? Is is a connectivity problem you could detect 
with something like "ping" or "Nagios?"

Since you say "different providers" I'm thinking a general connectivity problem 
or something "generally" out of whack with registrations.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] How to verify availability of the DID connection?

[Robert Augustyn]

Thanks 
 
Sincerely, 
Robert Augustyn 





From: da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On 
Behalf Of Danny Nicholas
Sent: Friday, March 06, 2009 9:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to verify availability of the DID connection?




Go to http://www.voip-info.org/wiki-Asterisk+tips+and+tricks and try some of 
the Dial Plan solutions.  You can probably find something to your liking that 
will work with little or no tweaking. 


From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn
Sent:Thursday, March 05, 2009 9:23 PM
To:asterisk-users@lists.digium.com
Subject:[asterisk-users] How to verify availability of the DID connection?


  

Hi all,


Occasionally, DIDs from different providers stop working for some reason.


I would like to be able to monitor situations like that and react before any of 
my clients start going ballistic on me.


Any ideas? Scripts you know of or wrote and willing to share?


Any info would be greatly appreciated.


  


Robert

  


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Re: [asterisk-users] How to verify availability of the DID connection?

[Robert Augustyn]

All these questions are valid, though I want first to see that the DID does not 
work then I will go and try to resolve it.  
I do not have a specific issue at this moment.

Sincerely,
Robert Augustyn

p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
 

-Original Message-
From: asterisk@sedwards.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, March 06, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to verify availability of the DID connection?

On Thu, 5 Mar 2009, Robert Augustyn wrote:

> Occasionally, DIDs from different providers stop working for some 
> reason.
>
> I would like to be able to monitor situations like that and react 
> before any of my clients start going ballistic on me.

Are you losing DIDs that terminate on your Asterisk box or your clients 
Asterisk box?

Are these DIDs registering with Asterisk and are you re-registering often 
enough?

Is it a problem within the providers? Can you port the DIDs to another provider?

Why do the DIDs stop working? Is is a connectivity problem you could detect 
with something like "ping" or "Nagios?"

Since you say "different providers" I'm thinking a general connectivity problem 
or something "generally" out of whack with registrations.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] How to verify availability of the DID connection?

[Robert Augustyn]

Hi all, 
Occasionally, DIDs from different providers stop working for some reason. 
I would like to be able to monitor situations like that and react before any of 
my clients start going ballistic on me. 
Any ideas? Scripts you know of or wrote and willing to share? 
Any info would be greatly appreciated. 
 
Robert 

 
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[asterisk-users] How to beep before transfer ...

[Robert Augustyn]

Hi,
When I transfer a call to an extension, the person I call does not have any 
idea when that transfer happened so it is a guessing game.
Is there a way to send a beep to the caller just before transferring the call? 
Preferably by setting something in FreePbx?
 
Sincerely, 
robert 




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[asterisk-users] Can I use an interact and visa terminal through VoIP?

[Robert Augustyn]

Hi, 
Is that reliable? Any known issues? or recommended setups? 
I am planning on adding the spa2002 devices and attaching the terminal to it. 
Will this work well? 
 
Sincerely, 
Robert Augustyn 



 
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[asterisk-users] Few of my phones do not ring when in a queue?

[Robert Augustyn]

Hi,
I have several Aastra 57i phones connected to 1.4.22 version of asterisk and 
when I call the queue these phones are part of I get few of these phones 
ringing with a delay ... as much as 18 secs or not at all ...
What could be the problem?
Thanks 

[Jan 22 03:06:19] VERBOSE[13842] logger.c: -- SIP/261-b7322c28 is ringing
[Jan 22 03:06:19] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-396e,1 is ringing
[Jan 22 03:06:19] VERBOSE[13841] logger.c: -- SIP/226-b7347828 is ringing
[Jan 22 03:06:19] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-b5f7,1 is ringing
[Jan 22 03:06:19] VERBOSE[13841] logger.c: -- SIP/226-b7347828 is ringing
[Jan 22 03:06:19] VERBOSE[13851] logger.c: -- SIP/233-b7324fb0 is ringing
[Jan 22 03:06:19] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-81a8,1 is ringing
[Jan 22 03:06:23] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:23] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-b410,1 is ringing
[Jan 22 03:06:25] VERBOSE[13838] logger.c: -- SIP/221-0944ee70 is ringing
[Jan 22 03:06:25] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-9761,1 is ringing
[Jan 22 03:06:25] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:25] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:25] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:25] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:25] VERBOSE[13843] logger.c: -- SIP/262-09457a68 is ringing
[Jan 22 03:06:25] VERBOSE[13834] logger.c: -- SIP/235-b7768038 is ringing
[Jan 22 03:06:25] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-0836,1 is ringing
[Jan 22 03:06:25] VERBOSE[13847] logger.c: -- SIP/229-b776bfb0 is ringing
[Jan 22 03:06:25] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-e77e,1 is ringing
[Jan 22 03:06:26] VERBOSE[13835] logger.c: -- SIP/236-b7310f70 is ringing
[Jan 22 03:06:26] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-af51,1 is ringing
[Jan 22 03:06:27] VERBOSE[13850] logger.c: -- SIP/230-b73511b0 is ringing
[Jan 22 03:06:27] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-b8dc,1 is ringing
[Jan 22 03:06:27] VERBOSE[13850] logger.c: -- SIP/230-b73511b0 is ringing
[Jan 22 03:06:37] VERBOSE[13840] logger.c: -- SIP/222-0944ee70 is ringing
[Jan 22 03:06:37] VERBOSE[13833] logger.c: -- 
Local/2...@from-internal-8853,1 is ringing

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[asterisk-users] How to monitor asterisk with SNMP?

[Robert Augustyn]

Hi, 
We have zabbix running and would love to be able to monitor our asterisk box 
with it. 
I believe that some sort of SNMP is build in 1.4+ correct? 
Where do I find more info or a how to on what is supported and how to use it? 
Thank you. 


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Re: [asterisk-users] Problems with sip registrations through HP Procurve 7102dl

[Robert Augustyn]

 
 It is not the source port being changed, it looks like the destination port is 
being changed.

robert
-Original message-
From: pe...@networkoblivion.com
Sent: Mon 29-12-2008 09:49
To: Asterisk Users Mailing List - Non-Commercial Discussion 
; 
Subject: Re: [asterisk-users] Problems with sip registrations through HP
Procurve 7102dl

Is it causing an issue?  There are lots of firewalls that do nat and 
change the source port of packets to some random udp port.  In my 
experience, for outbound registrations, it generally doesn't cause an issue.

Robert Augustyn wrote:
> Hi,
> I have a strange problem, when I try to connect to les.net from our 
> local asterisk server through Procurve router I seems to be connecting 
> on any port above 1024 and when I reload sip the port is changing too ...
> So I never get 5060? Any ideas on what is going on and how to resolve it?
>  
> Sincerely,
> Robert Augustyn
> 
> 519-997-3106 ext:802
> www.linqone.com <http://www.linqone.com/>
>   <http://www.linqone.com/>
>  
> 
> 
> 
> 
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[asterisk-users] Problems with sip registrations through HP Procurve 7102dl

[Robert Augustyn]

Hi,
I have a strange problem, when I try to connect to les.net from our local 
asterisk server through Procurve router I seems to be connecting on any port 
above 1024 and when I reload sip the port is changing too ...
So I never get 5060? Any ideas on what is going on and how to resolve it?
 
Sincerely, 
Robert Augustyn 

519-997-3106 ext:802 
www.linqone.com 
  

 
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[asterisk-users] How to disable trunk from the cli?

[Robert Augustyn]

Hi,
I need to be able to unable and disable iax2 trunks from the cli?
Is there a way to do it if so how?
 
Sincerely,
Robert Augustyn

 
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[asterisk-users] MoH in a loop

[Robert Augustyn]

Hi all,
Is it possible to have * playing an mp3 file in the way old tape system
worked?
 
 
Sincerely,
Robert Augustyn
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[asterisk-users] What is the best way to resale termination/origination?

[Robert Augustyn]

Hi,
We have been selling * systems for a while and always have used other
companies for origination and termination and let the client pay directly.
Since we do not have enough traffic to justify building our own
infrastructure we would like to start reselling someone else's service.
Any ideas? It would have to include easy way for billing. 
Thank you.
robert
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Re: [asterisk-users] Looking for a web video phone?

[Robert Augustyn]

Thank you,
How do I embed it into the web site though?
robert


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner
Sent: Monday, November 03, 2008 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Looking for a web video phone?










On Nov 3, 2008, at 12:11 PM, Robert Augustyn wrote:


Is there anything like that?
Any experiences?
 



X-Lite is a free download and has video capabilities. 



Fred Posner
[EMAIL PROTECTED]

Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999

www.teamforrest.com


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[asterisk-users] Looking for a web video phone?

[Robert Augustyn]

Is there anything like that?
Any experiences?
 
Sincerely,
Robert Augustyn

www.linqone.com <http://www.linqone.com/> 
 
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Re: [asterisk-users] Is anyone using * for 2 way video conferencing?

[Robert Augustyn]

Thank you.
What units from Polycom line did you use? 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Gordon Henderson
> Sent: Wednesday, October 29, 2008 4:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Is anyone using * for 2 way 
> video conferencing?
> 
> On Wed, 29 Oct 2008, Robert Augustyn wrote:
> 
> > Hi,
> > One of my clients, wants to use * box to run weekly 
> meetings between 
> > remote locations over the internet.
> > What would be the best configuration for this? We are talking about 
> > two conference rooms.
> > I am referring to the actual hardware/software and bandwidth 
> > requirements for this to work well.
> > I have run two software video phones and I had marginal 
> results with 
> > it when displayed on large LCDs, delay and blockines ware 
> the problems 
> > I have run into ...
> 
> I've been "playing" with video phones over the past month or 2.
> 
> You've got 3 choices: Bottom-end is Xlite, etc. soft-phones.
> 
> Desktop videophones - currently Grandtream GXV3000 and ATL4000's.
> 
> Top of the range Polycom video conferencing units.
> 
> Starting with the top-of the range ones - these "just work" 
> Don't even need an Asterisk box. Expensive though - I did one 
> help setup a pair of these, one in the UK, the other 
> west-coast US. Both with 42" plasma screens. Very nice, 
> worked very well. Very expensive.
> 
> More recently I've been using Grandstream GXV 3000's. For the 
> price; Fantastic. They do have audio and video outputs too - 
> I have connected one up to my 32" flat-screen TV and it 
> worked satisfactorily.
> 
> Picture quality is as good as the bandwidth you allow it to 
> use and they can go from 1 to 30 frames per second. It uses 
> about 128Kb/sec by default, but you can crank it up to 2 or 3 
> times that. The Polycoms I think were using about 225Kb/sec.
> 
> I've used the Grandstreamw with XLite - XLite using the same 
> codec, so same screen picture size. More or less just worked 
> when I got the codecs to match.
> 
> 
> So the big issue is the Internet - you're using a lot more 
> bandwidth, so need a better link. I found with the Polycoms 
> that the VPN we were using was introducing a lot of Jitter to 
> the link which degraded picture quality
> - turned off encryption and it was fine (cheap Draytek 
> routers doing encryption in software)
> 
> Right now, I'm using them in a more "domestic" setting than 
> business - I know more about the Internet in hte UK, so all 
> sites I'm experimenting with have good ADSL conections and 3 
> of us are on the same ISP, so minimising traffic over the 
> public Internet.
> 
> So there you go - hope this helps!
> 
> Gordon
> 
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[asterisk-users] Is anyone using * for 2 way video conferencing?

[Robert Augustyn]

Hi,
One of my clients, wants to use * box to run weekly meetings between remote
locations over the internet.
What would be the best configuration for this? We are talking about two
conference rooms.
I am referring to the actual hardware/software and bandwidth requirements
for this to work well.
I have run two software video phones and I had marginal results with it when
displayed on large LCDs, delay and blockines ware the problems I have run
into ... 
 
Sincerely,
Robert Augustyn
 <http://www.linqone.com>  
 
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Re: [asterisk-users] Cheapest 4 port FXO

[Robert Augustyn]

in multiport sipura/Linksys you cannot access individual ports you have to
address them by the group


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort
Sent: Sunday, October 26, 2008 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cheapest 4 port FXO


In this application what are the pros and cons of using a multiport ata vs a
tdm400/800/2400?

Eric


On Sat, Oct 25, 2008 at 10:54 AM, Joseph L. Casale
<[EMAIL PROTECTED]> wrote:


I need to increase reliability at an office as SIP/Internet provider outages
are causing some issues.
What would be the least expensive analogue card that people are using
reliably?

Thanks!
jlc
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Re: [asterisk-users] asterisk video

[Robert Augustyn]

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
> Sent: Thursday, October 23, 2008 6:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] asterisk video
> 
> 
> 
> Gordon Henderson wrote:
> > On Thu, 23 Oct 2008, Nhadie wrote:
> > 
> >> hi,
> >>
> >> hs anyone able to make video to work on asterisk? i tried 
> following this:
> >>
> >> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeB
> >> eam
> >>
> >> i can see that eyebeam is trying to broadcast a video but 
> the other 
> >> eyebeam is not receiving it.
> > 
> > What's the other end? Grandstreams won't take H263p, so force 
> > eyebeam/x-lite to only use H263, and just put this in as 
> the codec in 
> > sip.conf.
> 
> hi sir
> 
> both sides using eyebeam, i also hardset codec to use basic H236.
> 
> then i allowed the codec, allow=h236. i'm using Asterisk 
> 1.4.21.2., could that be the reason? coz document says video 
> on 1.4 is on infancy.

Make sure you have h263 and not h236 ...
I have the same configuration as yours and it work ok
If you still have problems post your relevant sip configuration

> 
> 
> > 
> > I presume you've read this too:
> > 
> >http://www.voip-info.org/wiki-Asterisk+video
> > 
> >> i tested the same setup but this time using ser with rtpproxy and 
> >> eyebeam video works fine.
> >>
> >> any ideas? where do you think should i start troubleshooting this?
> > 
> > I'm using Video with asterisk version 1.2 and it's working 
> very well 
> > so-far. Using Grandsteam phones - done one test with an ATL 
> phone and 
> > made a few calls to someone using x-lite. Tring to get 
> Ekiga to work, 
> > but I can't get the codecs for my wifes Acer One notebook.
> > 
> > Biggest problem is codec selection. Make sure both ends support the 
> > same thing, and that asterisk can route it.
> > 
> > Gordon
> > 
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Re: [asterisk-users] Ringtones for the console

[Robert Augustyn]

Thank you very much. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julien Claassen
> Sent: Thursday, October 09, 2008 4:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Ringtones for the console
> 
> Hi!
>One further notice about ringtone6: This existed long 
> before today. It's called "schon wie-der drei-zehn To-te", 
> the dashes are there to mark syllables. The translation is: 
> "Again 13 dead people". the the melody of a German 
> radiostation's traffic news. A German commedian came up with 
> the lyrics. :-) It just had to go in here. :-)
>Kindest regards
> Julien
> 
> 
> Music was my first love and it will be my last (John Miles)
> 
>  FIND MY WEB-PROJECT AT:  
> http://ltsb.sourceforge.net the Linux TextBased Studio guide 
> === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Anything what can be played through the console/dsp will work for me.
Yes, I received your application and hope to play with it tonight or
tomorrow.
Thank you very much. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julien Claassen
> Sent: Tuesday, October 07, 2008 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> No problem... I'll whomp something up. I'll upload a tarball 
> tomorrow or thrusday morning at the latest.
>Quality: desired samplingrate, bit-depth, channel number? 
> Any particular needs, or will CD quality just be fine for you?
>Kindest regards
>  Julien
> P.S.: Did you get to my application?
> 
> 
> Music was my first love and it will be my last (John Miles)
> 
>  FIND MY WEB-PROJECT AT:  
> http://ltsb.sourceforge.net the Linux TextBased Studio guide 
> === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Julien,
Thank you, I need a file which when played sounds like a phone ringing ...
:)
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julien Claassen
> Sent: Tuesday, October 07, 2008 3:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> Hello Robert!
>I don'texactly know, what you need for a "ringing file". 
> but if it is the matter of just some "announcement sound", I 
> could make you one. It's easy.
>Kindest regards
>  Julien
> 
> 
> Music was my first love and it will be my last (John Miles)
> 
>  FIND MY WEB-PROJECT AT:  
> http://ltsb.sourceforge.net the Linux TextBased Studio guide 
> === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Doug,
I have your example working but how do I get this to work with a ring group?
One more problem I have is poor quality of sound when the call file is
played.
I do not have this problem when moh is played or when console/dsp is used
for live voice?
What could be the problem?
Do you know where I can find a ringing file?
Thanks 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Doug Lytle
> Sent: Monday, October 06, 2008 4:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> Robert Augustyn wrote:
> > Ok then how do you make that an night_bell as your extension?
> >   
> 
> We have an after hours IVR, press 1 if you know the party 
> that you're trying to reach, press 2 for Dial By Directory 
> and press 3 for the night bell.
> 
> 
> [incoming]
> 
> ;
> ;* Check if call is within office hours,
> ;* if so, jump to the office-hours context
> ;* If not, continue on in the incoming
> ;* context.
> ;
> 
> exten => s,1,GotoIfTime(07:59-16:59|mon-fri|*|*?office-hours,s,1)
> exten => s,n,Answer()
> exten => s,n,Wait(1)
> 
> ;**
> ;* If after hours then play the 'Welcome'
> ;* and office hours message Press 1 if you know
> ;* the extension or 2 for dial by name directory
> ;**
> 
> exten => s,n,Background(local/welcome)
> exten => s,n,Background(local/business-hours)
> exten => s,n,Background(local/8am-5pm)
> exten => s,n,Background(local/press1-extension)
> exten => s,n,Background(local/press2-directory)
> exten => s,n,Background(local/press3-night-bell)
> 
> ;*
> ;* Set timeouts
> ;*
> 
> exten => s,11,Set(TIMEOUT(response)=15)
> exten => s,12,Set(TIMEOUT(digit)=2)
> 
> ;*
> ;* If 1 is pressed, go to Dial by extension
> ;*
> 
> exten => 1,1,Goto(dial-by-extension,s,1)
> 
> ;
> ;* If 2 is pressed, go to Dial by name
> ;
> 
> exten => 2,1,Goto(directory,s,1)
> 
> ;
> ;* If 3 is pressed, go to Night Bell
> ;
> 
> exten => 3,1,Goto(night_bell,4173,1)
> 
> 
> Doug
> 
> 
> -- 
>  
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a 
> little Temporary Safety, deserve neither Liberty nor Safety."
> 
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Julien,
I would love to see this solution so please upload the code.
Thank you very much.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julien Claassen
> Sent: Tuesday, October 07, 2008 4:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> Hi!
>I have a different approach. I wrote a small application 
> which simply starts an audio player. You can write a very 
> small script to answer fast or just use telnet like this:
> telnet localhost 8642
>At the moment everything is hardcoded, but can be changed 
> in any case. I use 15s ring-time, telnet-port 8642 and 
> mplayer as the audio-player.
>Short note: You don't have to submit anything over telnet, 
> just connect.
>If you're interested in this solution I'll upload the code 
> and give you a dialplan example (it's based in the return 
> code of the program.
>Kindest regards
>Julien
> 
> 
> Music was my first love and it will be my last (John Miles)
> 
>  FIND MY WEB-PROJECT AT:  
> http://ltsb.sourceforge.net the Linux TextBased Studio guide 
> === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Ok then how do you make that an night_bell as your extension?
Thanks 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Doug Lytle
> Sent: Monday, October 06, 2008 12:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> Robert Augustyn wrote:
> > Doug,
> > That is interesting concept.
> > How do you add this to a ring group and does it stop when 
> an extension 
> > is picked up?
> >   
> 
> It depends on how you have your ring group setup, I 
> personally only do this with a single extension.  And yes, 
> the bullhorn sound stops when the phone is answered.
> 
> Doug
> 
> 
> -- 
>  
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a 
> little Temporary Safety, deserve neither Liberty nor Safety."
> 
> 
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Re: [asterisk-users] PoE switch recommendations?

[Robert Augustyn]

Most phones support only 100M switching though  Unless you run separate
cabling for VoIP and data but then you would not need the 1G uplink. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> David Gibbons
> Sent: Monday, October 06, 2008 11:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PoE switch recommendations?
> 
> Obviously we don't need 1Gb connections for VOIP :)
> 
> Phones support pass through to the desktop and VLAN tagging.
> 
> The need for 1Gb ports comes from wanting to have 1Gb at the desktop.
> 
> Dave
> 
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Gordon Henderson
> Sent: Monday, October 06, 2008 11:29 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PoE switch recommendations?
> 
> On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:
> 
> > Hey, all.  We're rolling out VoIP, and I'm wondering about PoE 
> > recommendations, as we're going to have to replace our 
> current network 
> > equipment.  My first inclination would be to just plunk 
> down the cash 
> > and do a Cisco system, but I'm relatively certain that 
> would get shot 
> > down by finance.  Any recommendations for a couple-hundred-port 
> > solution with VLANs, PoE, and QoS?  Don't care much if it's in a 
> > single chassis or not, so long as it has Gbit uplinks.
> 
> I'm curious as to why you want Gb uplinks on the switches?
> 
> If we assume 100Kb/sec per phone .. (gross rounding, using 
> 100Kb/sec per phone, rather than ~80 - make the sums easier 
> and builds in a margin) 10 calls per Mb/sec.
> 
> So for a 24-port switch, 24 phones all talking to 24 
> extensions off that switch, the max the uplink port is going 
> to be pushing out is 2.4Mb/sec.
> 
> For 200 extensions, say 9 x 24 port switches, with a single 
> top-level (non PoE switch) switch with the PBX plugged in 
> along side the 9 downlinks, that single PBX link will be 
> carrying 2.4*9 = 22Mb/sec if all phones are in-use at the 
> same time (and the PBX is carrying media)
> 
> Now you may not want to build the network like that, but it 
> seems that Gb is overkill just for the VoIP side of things. 
> (And with that many extensions, I would suggest keeping all 
> the phones on one set of switches)
> 
> (Then again, it might not be possible to get big PoE switches 
> without Gb uplinks, so it might be a moot point!)
> 
> So satisfy my curiosity - why Gb uplinks?
> 
> Cheers,
> 
> Gordon
> 
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Doug,
That is interesting concept.
How do you add this to a ring group and does it stop when an extension is
picked up?
Thank you very much.
robert  

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Doug Lytle
> Sent: Monday, October 06, 2008 10:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to implement Ringing 
> through a sound card for overhead paging
> 
> Robert Augustyn wrote:
> > Hi,
> > I have followed this guide
> > 
> http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
> >  and have paging working ok, now I need to implement 'ringing'.
> > When someone calls I need the ringing to be send to overhead paging 
> > through the sound card.
> 
> I have recorded a sound effect, use a callfile to play the 
> file via the sound card.  I have a very short timeout for 
> that extension.  I just jump back to the beginning on the 
> context, play the sound effect and then ring the phone again. 
>  Code below:
> 
> 
> ;**
> ;* If Press extension is dialed after 5pm, play bull
> ;* Horn sound effect to get pressman's attention
> ;**
> 
> [night_bell]
> 
> exten => 
> 4173,1,GotoIfTime(07:45-16:59|mon-fri|*|*?press-officehours,s,1)
> exten => 4173,2,System(/bin/cp /usr/local/bin/bullhorn.call 
> /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten 
> => 4173,3,Dial(SIP/4173,15,tTkK) exten => 
> 4173,4,Goto(night_bell,4173,1)
> 
> 
> 
> 
> Channel: Console/dsp
> MaxRetries: 0
> Application: playback
> Data: /var/lib/asterisk/sounds/local/bullhorn
> 
> 
> Doug
> 
> -- 
>  
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a 
> little Temporary Safety, deserve neither Liberty nor Safety."
> 
> 
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[asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Hi,
I have followed this guide
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
 and have paging working ok, now I need to implement 'ringing'.
When someone calls I need the ringing to be send to overhead paging through
the sound card.
Any pointers?
 
 
Sincerely,
Robert Augustyn
www.linqone.com <http://www.linqone.com/> 
 
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[asterisk-users] Mexico Dids

[Robert Augustyn]

Hi,
I am looking for a did from Saltillo Mexico.
Any pointers?
robert
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Re: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem

[Robert Augustyn]

Load the sip on it and you good to go ... assuming the phones are ok ...


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dave cantera
Sent: Wednesday, December 19, 2007 12:27 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem



my client purchased a couple of shoreline ip-100 phones...  I managed to get
them to Not boot up...   shows the polycom logo then goes blank...   looks
like the want mcgp...  oh, mgcp...

is there a solution for this?  besides sending it back to polycom?
daveC





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Re: [asterisk-users] T.38 fax solution, opinions?

[Robert Augustyn]

How about connecting the fax machine directly to TDM card?
I was under the impression that this type of the solution is very solid ... 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Adam Moffett
> Sent: Monday, December 10, 2007 4:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T.38 fax solution, opinions?
> 
> If you mean faxing in audio it's hit or miss.  We do it here 
> and maybe have an error every 6 pages or so.  I wouldn't sell 
> it to a customer as a solution.
> > How about fax machines talking directly to spa2102 and then out the 
> > pri or am I missing something?
> >
> > 
> > 
> --
> > --
> >
> 
> 
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Re: [asterisk-users] T.38 fax solution, opinions?

[Robert Augustyn]

How about fax machines talking directly to spa2102 and then out the pri or
am I missing something?


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson
Sent: Monday, December 10, 2007 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T.38 fax solution, opinions?



I;m not sure how your solution would work. but I thought I'd throw out some
ideas that we are having to implementing faxing here on a new install.

 

We are going to be bringing in a PRI and routing all the DIDs from our
existing copper lines to the PRI (including fax DIDs). the solution we are
working towards is certainly not ideal, but we are hopeful its going to
work.

 

Incoming fax calls will come into Asterisk, asterisk will route them to
IAXmodem which will feed Hylafax (all running on the same box as Asterisk to
reduce latency. and we are talking about fairly small volumes).  Hylafax
will then do fax 2 email

 

Outbound faxing however is a bit trickier.

 

We are going to use Linksys ATAs (tested with a SPA2102) which will have the
POTS fax machines plugged into them, the SPA2102 connects to asterisk with
SIP, asterisk will then route the calls to t38modem (recent dev versions of
it support SIP and not only H.323), t38modem is basically just like IAXmodem
except its SIP and supports t38 termination.  T38modem will again feed
hylafax, which will then route back to asterisk through IAXmodem and then up
the PRI.

 

Its certainly not a pretty solution. but we haven't come up with anything
else yet.. the only step I can see possibly simplifying is on outbound faxes
Hylafax can possibly be bypassed and have t38modem talk directly to
IAXmodem.

 

--

Matt

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of arkda
Sent: Sunday, December 09, 2007 11:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] T.38 fax solution, opinions?

 

Hi,

I'm putting together a fax solution for my company that utilizes T.38. I
wanted to throw out my plan and get some feedback if anyone is doing
something similar or sees a blatant problem with it.

We're currently rolling out SPA-942 phones for the standard desk phone with
vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls
for satellite offices are handled by VoIP providers (for voice Vitelity
inbound, Gafachi outbound). These satellite offices are using a T.38 fax DID
from Gafachi, passed through the Asterisk server to a Linksys 3102 ATA and
then to a POTS fax machine. This all works well thus far. 

Our HQ has a full voice PRI, terminated on the Asterisk server with a
TE120P. Additionally, right now they have five fax lines totally separate
from the PRI that are used for POTS fax machines.

I'm thinking of porting those numbers to the PRI and purchasing a TDM880B
(comes with eight FXS modules) and routing the fax DIDs to the 880 in
Asterisk. Five of the ports would connect into a Linksys 3102 that would
speak T.38 to what would be our new fax environment (Exchange 2007 Unified
Messaging). That part isn't implemented yet, but it shouldn't be a problem.
Once it's implemented I'll probably reroute the Gafachi T.38 fax DIDs to
Exchange through Asterisk (with sipX in there somewhere).

The part(s) I'm unsure about is the TDM880B. I haven't used a FXS card with
Asterisk, and I certainly haven't used a fax machine on that FXS.
Additionally, I'm not 100% sure the 3102 will talk directly to Exchange UM
yet, but that's something I can figure out myself soon; I'm just not sure
about spending the cash for a TDM880B without knowing someone has thrown
faxes through it from a PRI terminated on the same box from a separate card.


Anyway, thoughts, criticisms, insults and stinging barbs all welcome.

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[asterisk-users] Testers needed for VoIP router solution

[Robert Augustyn]

Hi all,
We have put together a firmware for a range of inexpensive routers.
It has been configured to provide optimum VoIP performance.
We have internally tested it for number of months and it looks very good.
You should be able to run it easily with 20+  phones on local network ( we
still did not hit the upper limit ) assuming that you have bandwidth.
Your VoIP will get prioritized over other types of traffic.
You should be able to talk, download files and run torrents at the same time
with no visible degradation of the VoIP voice quality.
It will be delivered ready to upload with all your configurations, which you
will have to provide to us.
We will custom build firmware for your configuration.
We just ask you to upload it, test it and provide feedback.
 
If you are interested ( sorry only first 10 will be accepted ) please
contact me at firmware at linqone dot com and we will send you the set of
questions we need you to answer before we can build a solution for you.
Thanks,
 
 This firmware will work on: 

*   Linksys WRT54G v1-v4, WRT54GS v1-v4,
<http://www.amazon.com/gp/search?ie=UTF8&tag=jon002&index=blended&linkCode=u
r2&camp=1789&creative=9325&keywords=linksys%20wrt> WRT54GL v1.x,
<http://www.amazon.com/gp/search?ie=UTF8&tag=jon002&index=blended&linkCode=u
r2&camp=1789&creative=9325&keywords=linksys%20wrtsl54gs> WRTSL54GS (no USB
support) 

*   Buffalo
<http://www.amazon.com/gp/search?ie=UTF8&tag=jon002&index=blended&linkCode=u
r2&camp=1789&creative=9325&keywords=buffalo%20whr> WHR-G54S,
<http://www.amazon.com/gp/search?ie=UTF8&tag=jon002&index=blended&linkCode=u
r2&camp=1789&creative=9325&keywords=buffalo%20whr> WHR-HP-G54, WZR-G54,
WBR2-G54 

*   Asus
<http://www.amazon.com/gp/search?ie=UTF8&tag=jon002&index=blended&linkCode=u
r2&camp=1789&creative=9325&keywords=asus%20wl500g%20premium> WL500G Premium
(no USB support) 

This will not work on Linksys WRT54G/GS v5-v7 or newer WRT54G/GS routers.
 
If you do not have any of the above routers you can get one for UNDER $40
shipped at:
 
 
<http://www.circuitcity.com/ccd/Search.do?c=1&context=&keyword=Buffalo+WHR-G
54S&searchSection=All&go.x=11&go.y=10>
http://www.circuitcity.com/ccd/Search.do?c=1&context=&keyword=Buffalo+WHR-G5
4S&searchSection=All&go.x=11&go.y=10 
 
How do I find my Linksys WRT54G/WRT54GS/WRT54GL's version?
Look at the bottom side of the router to check for the version number, or
compare the first 4 characters of the serial number with the following list:

CDF0/CDF1 = WRT54G v1.0
CDF2/CDF3 = WRT54G v1.1
CDF5 = WRT54G v2.0
CDF7 = WRT54G v2.2
CDF8 = WRT54G v3.0
CDF9 = WRT54G v3.1
CDFA = WRT54G v4.0

CGN0/CGN1 = WRT54GS v1.0
CGN2 = WRT54GS v1.1
CGN3 = WRT54GS v2.0
CGN4 = WRT54GS v2.1
CGN5 = WRT54GS v3.0
CGN6 = WRT54GS v4.0

CL7A = WRT54GL v1.0
CL7B = WRT54GL v1.1


If it's not listed above, and it's not a WRT54GL, it's not supported. 
 
 
 
Sincerely,
Robert Augustyn
 
This firmware is provided as-is without any warranty. I will NOT be
responsible for damages that occur due to the use of this firmware. USE AT
YOUR OWN RISK.
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[asterisk-users] CITEL gateway does it work well?

[Robert Augustyn]

Hi all,
Is using a Citel gateway with Asterisk a good solution for reusing of the
old Nortel digital phones?
Would love to get some input from actual users.
Any/all opinions welcome.
robert
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RE: [asterisk-users] Re: RE: Digital Phones

[Robert Augustyn]

Stephen,
I understand that these sets are digital but what about connecting  Asterisk
fxs to Nortel fxo and keep sets connected to existing Nortel?

Would that work?
Robert

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Stephen Bosch
> Sent: Wednesday, May 09, 2007 8:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Re: RE: Digital Phones
> 
> Robert Augustyn wrote:
> > Can you connect existing Nortel system to Asterisk through fxs/fxo?
> > That way one could use existing infrastructure for few old 
> phones and 
> > Asterisk for new phones and all good things which come with it?
> 
> No. They are digital phones and use proprietary Nortel signalling.
> 
> The "too bad" thing about Nortel is that, when all your 
> infrastructure is Nortel, it's pretty solid, reliable stuff, 
> and in its day it was also pretty amazing. There's a reason 
> why their PBX hardware was the most widely deployed in North America.
> 
> Every technology has its day, though, and Nortel has been 
> milking its contribution since 1975. The new IP stuff is 
> unimpressive; I considered getting BCM certification once but 
> when I looked at the equipment costs, I just shuddered. No 
> sane business operator would pay those prices, and most of 
> the insane ones are already in jail.
> 
> Your best bet is to sell the sets you already have and 
> replace them with appropriate IP hardware (you might even 
> consider Aastra, which inherited most of Nortel's 
> conventional telephony portfolio and has done great things 
> with it); you can still get good money for used Nortel 
> digital sets; many people are perfectly happy with their 
> systems and I expect they'll be around for some time to come.
> 
> -Stephen-
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RE: [asterisk-users] Re: RE: Digital Phones

[Robert Augustyn]

Can you connect existing Nortel system to Asterisk through fxs/fxo?
That way one could use existing infrastructure for few old phones and
Asterisk for new phones and all good things which come with it?
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Eric "ManxPower" Wieling
> Sent: Wednesday, May 09, 2007 1:28 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Re: RE: Digital Phones
> 
> bilal ghayyad wrote:
> > Hi;
> > 
> > Well, I understood now that Nortel has some digital phones 
> that can be 
> > used with astrisk, but the
> > question: what are the card models that should be installed on 
> > Asterisk server? Digium? What these models?
> 
> None.  There are no Nortel digital phones that work with 
> Asterisk.  As I understand it, they MAY have some SIP phones, 
> but I suspect they use a Nortel variant of SIP.
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RE: [asterisk-users] Developing Marketing materials ...

[Robert Augustyn]

Dave,
Agreed, is there anything you have in mind?
robert


  _  

From: dave cantera [mailto:[EMAIL PROTECTED] 
Sent: Saturday, April 21, 2007 5:39 PM
To: Robert Augustyn
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion';
[EMAIL PROTECTED]
Subject: Re: [asterisk-users] Developing Marketing materials ...


robert, matt,
sounds good...  I am looking for a high quality series of pieces that are
both informative and strike a point.  by quality, I am not talking about
paper and layout... it is the message that I am looking for the paper,
pictures, typeface are just vehicles to get the reader interested to take
the next step...
thanks for taking point on this project,
daveC

Robert Augustyn wrote: 

Matt and Dave,

I am open to all suggestions and solutions but if we have few people

involved then cost stops being an issue ( to some extend ).

I want this to be done well as we need an edge.

I will let it run for few days and then contact all who expressed interest

on the board or directly to me.

Thank you.

robert 



  

-Original Message-

From: [EMAIL PROTECTED] 

[mailto:[EMAIL PROTECTED] On Behalf Of 

Matt Gibson

Sent: Saturday, April 21, 2007 2:59 PM

To: [EMAIL PROTECTED]; Asterisk Users Mailing List 

- Non-Commercial Discussion

Subject: Re: [asterisk-users] Developing Marketing materials ...



I am also working on this, and have a 

marketing/communications background. I may be able to help 

cheaper than the "big agency" :)



thanks,

matt





On 20/04/07, dave cantera  <mailto:[EMAIL PROTECTED]>
<[EMAIL PROTECTED]> wrote:



robert,

I might be interested depending on cost, message, and quality...

keep me in the loop.

daveC



Robert Augustyn wrote:

  

Hi,

I am working on developing a professional Marketing 



Materials for my 



systems.

I plan on using a very good(expensive) company to do that so 

splitting the costs with several people would be nice.

Let me know if you are interested on taking part in it.

robert















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No virus found in this incoming message.

Checked by AVG Free Edition.

Version: 7.5.446 / Virus Database: 269.5.1/765 - Release Date: 

04/17/2007 05:20 PM





--

Building Strong Relationships w/ Intelligent Customer Service

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Interlocking Business Solutions, LLC

856-380-0894 x5000





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856-380-0894 x5000



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RE: [asterisk-users] Developing Marketing materials ...

[Robert Augustyn]

Matt and Dave,
I am open to all suggestions and solutions but if we have few people
involved then cost stops being an issue ( to some extend ).
I want this to be done well as we need an edge.
I will let it run for few days and then contact all who expressed interest
on the board or directly to me.
Thank you.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matt Gibson
> Sent: Saturday, April 21, 2007 2:59 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List 
> - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Developing Marketing materials ...
> 
> I am also working on this, and have a 
> marketing/communications background. I may be able to help 
> cheaper than the "big agency" :)
> 
> thanks,
> matt
> 
> 
> On 20/04/07, dave cantera <[EMAIL PROTECTED]> wrote:
> > robert,
> > I might be interested depending on cost, message, and quality...
> > keep me in the loop.
> > daveC
> >
> > Robert Augustyn wrote:
> > > Hi,
> > > I am working on developing a professional Marketing 
> Materials for my 
> > > systems.
> > > I plan on using a very good(expensive) company to do that so 
> > > splitting the costs with several people would be nice.
> > > Let me know if you are interested on taking part in it.
> > > robert
> > >
> > > 
> 
> > > 
> > >
> > > ___
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> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > 
> 
> > > 
> > >
> > > No virus found in this incoming message.
> > > Checked by AVG Free Edition.
> > > Version: 7.5.446 / Virus Database: 269.5.1/765 - Release Date: 
> > > 04/17/2007 05:20 PM
> > >
> >
> > --
> > Building Strong Relationships w/ Intelligent Customer Service
> > --
> >
> > Interlocking Business Solutions, LLC
> > 856-380-0894 x5000
> >
> >
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[asterisk-users] Developing Marketing materials ...

[Robert Augustyn]

Hi,
I am working on developing a professional Marketing Materials for my
systems.
I plan on using a very good(expensive) company to do that so splitting the
costs with several people would be nice.
Let me know if you are interested on taking part in it.
robert
 
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RE: [asterisk-users] GSM cleanup (pops, clicks and static)

[Robert Augustyn]

Audacity  

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> David Ruggles
> Sent: Friday, February 23, 2007 4:48 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] GSM cleanup (pops, clicks and static)
> 
> I have a bunch of sounds that I've converted into gsm from 
> (Indexed NMS) vox files. There's only a single utility that 
> I've found that can read and convert vox files. My conversion 
> process is to use this utility to convert the index vox file 
> in to a series of wave files and then use sox to convert the 
> wave files to gsm files. Over all this works really well, the 
> problem is that about 60 to 70 percent of the gsm files have 
> some static or popping and clicking, on most of them it is in 
> the silence at the end of the file.
> 
> All that back story to ask this question: Are there any good 
> utilities available for cleaning up gsm files?
> 
> Thanks,
> 
> David Ruggles
> CCNA MCSE (NT) CNA A+
> Network Engineer  Safe Data, Inc.
> (910) 285-7200[EMAIL PROTECTED]
> 
> 
> 
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[asterisk-users] Should I use sip gateway of PCI card?

[Robert Augustyn]

Hi,
I am planning couple small business installations and wader what should I
use for 2 to 6 lines a gateway or pci card?
Any comments greatly appreciated on pros and cons and brands.
Thanks,
robert

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RE: [asterisk-users] Best inexpensive home office router for VoIP(QoSwith maybe PoE)

[Robert Augustyn]

Open-wrt on supported router and some custom scripting works very well. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> Sent: Thursday, January 04, 2007 3:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Best inexpensive home office 
> router for VoIP(QoSwith maybe PoE)
> 
> Yes, I knew that but it's nice that you mention it.  I want 
> QoS specifically to prevent large downloads/kids using 
> BitTorrent in their bedrooms locally from interfering with the calls.
> 
> Mike
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> Sent: Thursday, January 04, 2007 13:32
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Best inexpensive home office 
> router for VoIP(QoS with maybe PoE)
> 
> Mike wrote:
> > Hi,
> >  
> > I'm looking for opinions on the "best value" router to use for home 
> > offices.  It should work for a scenario in which there are 
> 3 computers 
> > and 2 SIP phones, handling QoS so that the phones always 
> have higher 
> > priority traffic than the PCs. (and not rely on the phones 
> to do the 
> > QoS because some PCs may not be connected to the phones).
> >  
> > QoS could be based on destination and source IP (i.e. an Asterisk
> > server) or MAC address of the phones. Ideally with PoE, but at this 
> > point it's just a bonus.
> >  
> > What are people on this list using?  I've found that the 
> mention QoS 
> > on a box doesn't guarantee any real QoS functionality.
>  
>  QoS on the router will only guarantee your phone traffic 
> gets higher priority than other (web, mail etc..) however 
> once the bits leave your router any QoS is essentially lost 
> as the call traverses the internet.
> Having QoS on your router is valuable to prevent some large 
> download from buggering your calls though.
> 
> 
> 
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RE: [Asterisk-Users] About Softphone IAX free for Pocket PC

[Robert Augustyn]

Steve,
Sorry, it is not iaxcomm it is :http://www.voipalia.com/ppciax/
Good luck.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steve Jones
> Sent: Wednesday, April 26, 2006 10:49 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
> 
> I can't find a pocket pc version of that on the iaxcomm 
> website..  Only linux, Mac, Windows..  Can you send a link?  
> This is exactly what I'm looking for!!  Thanks!
> 
> -Original Message-
> From: Robert Augustyn [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, April 25, 2006 12:38 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
> 
> I use IaxComm with good results on axim x51 
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RE: [Asterisk-Users] About Softphone IAX free for Pocket PC

[Robert Augustyn]

Steve,
I a sorry, I should have verified what I am writing.
The software is PPCIAX2 and you can find it: http://www.voipalia.com/ppciax/
Is it pretty not but it works.
robert
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steve Underwood
> Sent: Tuesday, April 25, 2006 12:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] About Softphone IAX free for Pocket PC
> 
> Robert Augustyn wrote:
> 
> >I use IaxComm with good results on axim x51
> >  
> >
> Is that something you developed yourself? If so, can you 
> share it? For the last year I have been trying to find time 
> to get iaxcomm working on a WinCE machine.
> 
> Regards,
> Steve
> 
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RE: [Asterisk-Users] About Softphone IAX free for Pocket PC

[Robert Augustyn]

I use IaxComm with good results on axim x51 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Kerry Garrison
> Sent: Tuesday, April 25, 2006 11:26 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] About Softphone IAX free for Pocket PC
> 
> 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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> 
> 
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[Asterisk-Users] RE: Is my math on traffic/bandwidth correct?

[Robert Augustyn]

 Hi,
 According to what I read, for g.729A 1 line I need 21 kbps.
 Now is it 21 coming in and 21 going out? Or 10.5 coming in 
 and 10.5 going out?
 Assuming that it is 21 coming in my traffic would be 
 21x3600=75600kb = 9450KB /per hour So close to 10M of traffic 
 is coming into my network per hour per 1 g.729 line.
 Is that close or am I completely off 
 Thanks,
 robert
 

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RE: [Asterisk-Users] VOIP Router

[Robert Augustyn]

Arek,
Where can you get these?
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Arek Bekiersz
> Sent: Thursday, January 26, 2006 7:50 AM
> To: [EMAIL PROTECTED]
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] VOIP Router
> 
> Hi,
> 
> 
> Try one of Venus 2804, 2808 or 2832 from Tainet corporation.
> They support SIP or MGCP and they come with VPN.
> 
> http://www.tainet.net
> Proceed to "Product/VoIP/Venus"
> 
> --
> Regards,
> Arek Bekiersz
> 
> 
> 
> Mohamed Farid wrote:
> > Dear All :
> > I need to link my HQ to some Remote Sites - I need a Router which 
> > supports VOIP , and VPN Also the Router Should has 3 FXS 
> ports and 1 
> > FXO ...
> > The call should be routed from the Remote Site to the HQ 
> through a VPN 
> > tunnel ( 3DES ) ...
> > Any Advise ?
> > Mohamed Farid ,,
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Robert Augustyn]

Are GSM gateways allowed in Canada?
And can we resell it?
Robert


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steve Kennedy
> Sent: Friday, January 06, 2006 9:17 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale
> 
> On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
> 
> > > I don't get it. What is the advantage of using a GSM gateway? 
> > > VOIP calls are pretty inexpensive as they are now.
> > It largely depends on the country you're calling. Here in the UK, 
> > calls to mobiles are maintained at an artificially high 
> rate because 
> > the terminating network (the mobile networks) get a cut of call 
> > revenue for calls *to* your mobile. By contrast, in the US, 
> the mobile 
> > customer often pays a small charge per minute on incoming 
> calls (as I understand the market over there).
> > You'll also find in the UK the mobile phone market is heavily 
> > subsidized by the networks such that you can get phones for free if 
> > you sign up to 12 month contracts. I often find that it's 
> > cost-effective to get a new contract every 12 months (with a free 
> > phone), even if I don't want the phone. Flog the phone on ebay and 
> > you've got a spare SIM with lots of inclusive minutes for 
> almost nothing.
> 
> In the UK the wholesale rates are set by Ofcom (like the 
> FCC), which works out about 7p'ish per minute.
> 
> However the operators can offer retail bundles (including 
> phones) and for a monthly contract they "throw" in various 
> ammounts of cross network minutes (or free to their own 
> network or whatever). With clever dial-plans and multiple 
> terminals connected to multiple networks you can generally 
> get "free" calls to mobile users (basically clever least cost 
> routing, time of day sometimes needs to be taken into account 
> as well).
> 
> However there are some disadvantages, the main being you cant 
> set CLI of the outgoing call as it will always be tied to the 
> SIM of the mobile terminal.
> 
> Another is that you can NOT run a GSM gateway (as they're 
> known) for 3rd parties. So if you want to connect your office 
> PBX to a gateway to make use of cheap mobile termination for 
> your own company that's fine, but as an ITSP (or traditional 
> telco) you can not allow 3rd party traffic to utilise a 
> gateway. If networks find you are using a gateway (as a 
> telco) they can cut it off, no questions asked. Gateways have 
> been determined to be fixed infrastructure, therefore NOT mobile.
> 
> There is (or maybe was by now) an Ofcom consultation asking 
> whether this should be changed, the mobile operators will 
> fight it, telcos and other users will be asking for it to be changed.
> 
> Of course this is UK specific, other countries have more 
> lenient policies (I think Belgium allow gateways, France 
> doesn't allow any kind, and some allow them with the 
> co-operation of the operators).
> 
> 
> Steve
> 
> --
> NetTek Ltd  UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 
> 7483 2455 Skype/GoogleTalk/AIM stevekennedyuk / MSN 
> [EMAIL PROTECTED] Euro Tech News Blog 
> http://eurotechnews.blogspot.com 
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Robert Augustyn]

What is the price and availability?
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Cory Andrews
> Sent: Thursday, January 05, 2006 3:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale
> 
> SICPE has a new product called the GSM Call Director that may 
> be of interest to GSM enthusiasts.
> 
> http://www.sipcpe.com/fx300GSM.html
> 
> Cory Andrews
> Purchasing Manager
> ++
> VOIPSupply.com
> A Division of b2 Technologies
> 454 Sonwil Drive
> Buffalo, NY 14225
> 
> direct - 716.250.3402
> mobile - 716.907.4054
> email - [EMAIL PROTECTED]
> AIM - b2Cory
> 
> - Original Message -
> From: "Sam Tam" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Sent: Thursday, January 05, 2006 3:30 PM
> Subject: RE: [Asterisk-Users] GSM Gateway / Terminal for sale
> 
> 
> We have ran out of stock in our office in UK. All GSM Gateway 
> are now being
> send from HK therefore the shipping will be more expensive than usual.
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of bails
> Sent: Friday, January 06, 2006 12:18 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale
> 
> Chris Bagnall wrote:
> >>Single port GSM Gateway support 900 / 1800 GSM mode with
> >>external antenna.
> >>Brand new unit and all of them will be tested before dispatch.
> >>Extremely easy to setup and can be used out of the box
> >>without any configuration. So should be good alternatively of
> >>phonecell or nokia pbx etc..
> >>Units are located in UK and £60 GBP per unit excluding shipping.
> >
> >
> > Has anyone bought one of these and able to offer some feedback? I'm
> > seriously considering a GSM gateway to take advantage of 
> the spare SIM
> cards
> > lying around still inside their 12-month contracts.
> >
> > Looking at the website in question, delivery is £17.37 for a 6-day
> delivery,
> > or £10 for a 30+ day delivery, both of which seem a bit 
> high for an item
> > apparently located in the UK.
> >
> > Regards,
> >
> > Chris
> 
> We were working in the area (Reading) and offered to pay cash and
> collect from their site, but the response was;
> 
> "that they could only be sent direct from the far east"
> 
> We weren't prepared to take the risk, I mean they turned down cash!
> 
> Bails
> 
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> 
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RE: [Asterisk-Users] Linksys SPA-9000

[Robert Augustyn]

Where can I get more info on this product?
Thanks 
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Cory Andrews
> Sent: Thursday, December 29, 2005 1:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Linksys SPA-9000
> 
> Kerry - we certainly have the ability to match any pricing 
> you are receiving from Atacomm, we offer discounted VAR 
> pricing but you need to have an account set up with an inside 
> sales rep, and I can facilitate that.  We also offer RMA 
> support and firmware for the duration of manufacturer 
> warranty on all the products we sell, whereas Atacomm only 
> offers a 30 day warranty.
> 
> We will be offering the SPA-9000, and are looking to contract 
> with a few SP's, so we can upsell voicemail provisioning in 
> conjunction with the units.  We have the ability, but not the 
> desire, to host the VM internally, as we are not a service 
> provider, and don't want the potential headaches that come 
> along with that.  We may simply decide to act as a 
> fulfillment and marketing agent for an SP, or group of SP's, 
> and when we sell the SPA-9000, we are selling it on your 
> behalf.  We will likely offer pre-provisionment of the unit 
> for customers that want that, and many customers don't want 
> to deal with configuring the system even though it is fairly 
> straightforward. 
> 
> We could probably set up some automated provisioning setup, 
> so that you could remotely provision the SPA-9000 to the 
> clients spec, and then we package and ship it.  We can also 
> do the provisionment in house, we provide outsourced SIP 
> device provisionment and fulfillment for a wide variety of 
> VOIP service providers, including DeltaThree, iConnectHere, 
> Broadvoice, and hopefully soon, Vonage.
> 
> We have a web based form where a client can outline how they 
> want their autoattendant(s), extensions and other options 
> configured.  We have voice talent for prompts, or clients can 
> provide their own.  We offer a wide spectrum of value adds, 
> including leasing and finance options, because as you know 
> there is not a ton of margin in this hardware.
> 
> If you have some time to chat later this week, I am anxious 
> to see how SP's foresee pricing the voicemail service for the 
> SPA-9000, on a "per seat" model, or a "per pbx" model.
> 
> Let me know a convenient time to reach you.  We are certainly 
> looking for strong partnerships and we bring a lot to the 
> table, with sales approaching $50MM and aggregating, on 
> average, around 2200 new customers per month over the past year.
> 
> Thanks for the email!
> 
> Regards,
> 
> Cory Andrews
> Senior Partner
> +++
> VOIPSupply.com
> 454 Sonwil Drive
> Buffalo, NY 14225
> +++
> voice - 716.630.1555 X22
> email - [EMAIL PROTECTED]
> fax - 716.630.1548
> 
> 
> 
> Kerry Garrison wrote:
> 
> >Cory,
> >  Sherman at Linksys suggested I touch bases with you. We have an 
> >SPA-9000 here that we are testing out. We will be rolling out a 
> >voicemail service to go along with it as well. We have a 
> small IT firm 
> >in southern California and are growing our IP PBX business 
> quite nicely 
> >this year and expect 2006 to be very nice. We have been 
> buying strictly 
> >from atacomm because of their prices but would rather have a good 
> >partnership with someone that can potentially help us out as 
> well if you need help on the west coast.
> >
> >Just thought I would make the introduction and see if we can start 
> >talking about how we can work together.
> >
> >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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> >
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RE: [Asterisk-Users] asterisk in real estate developments

[Robert Augustyn]

Colin,
Nice summary, what gateway are you using and with what carrier.
Thanks,
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Tuesday, December 13, 2005 10:53 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] asterisk in real estate developments
> 
> been there done that
> 
> The biggest problems that we have had to deal with is 
> stability. Stability with respect to system stability and 
> also stability of the user.  System stability is extremely 
> difficult to deal with when you have poor power, power 
> outages, guys pressing the buttons next to the blinkenlights 
> because they are curious (very difficult to control access to 
> gear in a construction trailer when everything is set up 
> ad-hoc) and users doing stupid things like moving their phone 
> from their desk to another guys desk for god-knows-why reason 
> then not plugging into the cat5 cable. Then the users tend to 
> say that your phone system is shit, when these types of 
> problems are largely beyond your control. 
> 
> As to user stability, I don't refer to mental stability, I 
> refer to churn.
> Staff churn in the construction industry is brutal and 
> combined with the relative unsophistication of people in the 
> construction industry, it's the Training Session That Will Never End. 
> 
> My evolution of how to deal with this problem has gone from 
> broadband to a construction trailer, wifi and wired sip 
> phones, to masking a user's cell number behind a DID and 
> cutting airtime with a GSM gateway. Really, it's the best 
> way, treating cell phones as "extensions". Masking the cell 
> number behind a DID allows me to get the audio inside of 
> Asterisk so I can do VoIP-y things with it, the GSM gateway 
> adds very little incremental cost (a
> 4 port GSM gateway adds only, for us, $100 Cdn a month to our 
> cell costs and saves us $2-3k a month at 25c / min) and 
> there's not too much of a training issue, since these 
> blockheads grock their cell phones already. We even have
> 4 digit extension dialling from office staff to cells, and 
> call transfer from the cell. MWI and inbound fax reception 
> notification is done via SMS (each DID can recieve faxes, and 
> the fax goes to the user's email, thanks SpanDSP!). Overflow 
> when the gateway is full goes out our PRI, and we eat the 
> airtime, but that's a good thing because if the GSM gateway 
> is full, we are saving money. 
> 
> -Original Message-
> From: Steve Totaro [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 13, 2005 8:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] asterisk in real estate developments
> 
> 
> Very true.  How about using WiFi and DEC phones?
> 
> Thanks,
> Steve
> 
> > 
> >  hey chris,
> > 
> > The only issue you'll run into is that with all temp stuff like 
> > construction trailers ect they like to cut there lines A 
> LOT  with all 
> > there nice machines.
> > 
> > Carlos Alcantar
> > Race Technologies, Inc.
> > 101 Haskins Way
> > South San Francisco, CA 94080
> > P: 650.246.8900
> > F: 650.246.8901
> > E: [EMAIL PROTECTED]
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Chris 
> > Bagnall
> > Sent: Tuesday, December 13, 2005 5:08 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] asterisk in real estate developments
> > 
> > > I was wondering if anyone has used asterisk in a real estate 
> > > development project. I know someone that is developing a 
> ~400 home 
> > > project and thought asterisk might be a possible 
> alternative to the 
> > > phone company and a way to offer more service to buyers.
> > 
> > How about deploying asterisk to support the contractor 
> responsible for
> the
> > construction of these sites? Instead of developers (who are often
> on-site
> > for 6 months plus) relying purely on cellphones or asking 
> the ILEC to 
> > install a load of phone lines for them, stick an asterisk server in
> their
> > site office linked to a net connection, shove a load of cordless
> phones on
> > a
> > channel bank at convenient points around the site and 
> contractors are 
> > never far from a phone.
> > 
> > This is something we're hopefully doing for a property developer in
> the
> > new
> > year. It'll be interesting to see how well it all works out.
> > 
> > Regards,
> > 
> > Chris
> > --
> > C.M. Bagnall, Director, Minotaur I.T. Limited This email is 
> made from
> 100%
> > recycled electrons
> > 
> > 
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RE: [Asterisk-Users] Messages button on a Polycom 501

[Robert Augustyn]

Add this to your phone1.cfg:



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="" 
My extension is 805 so you will 
be sending *98805 to get to your voicemail from there you may need to enter your 
password or not ...:)
Works like a charm 
..
robert

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Brent 
  BloodworthSent: Tuesday, December 06, 2005 11:30 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Messages button on a Polycom 501
  I mistakenly followed a how to guide on voip-info.org describing how to setup the 501s 
  with [EMAIL PROTECTED] It appears that you have set me on the right track as 
  setting the contact set to *98 brings up the voicemailmain. The next logical 
  question is - How do I setup the contact to enter the mailbox number and 
  password when prompted. I would assume that this can be  scripted but I 
  am a noob when it comes to asterisk. thanks for all the help so far. 
  On 12/6/05, Eric 
  ManxPower Wieling <[EMAIL PROTECTED]> wrote:
  You 
have the contact set to the extension, you need the contact set 
towhatever you dial to retrieve your voicemail.  i.e. the one 
that runsvoicemailmain.Brent Bloodworth wrote:> Actually 
I think that is how it is setup now. I configured the phone > through 
the web interface. Callback mode is set to "contact" and the> 
Callback contact is set to the extension.>> On 12/5/05, *Jerry 
Jones* <[EMAIL PROTECTED] 
[EMAIL PROTECTED]>>> 
wrote:>> I assume you wish to have the 
button retrieve your vm - if so then>> 
time to edit your config file, or use web interface. 
>>   msg.mwi.1.subscribe="" 
msg.mwi.1.callBackMode="contact" msg.mwi.> 
1.callBack="xxx"> xxx=extension to dial to 
retrieve vm>> On Dec 5, 2005, at 5:38 
PM, Brent Bloodworth wrote: 
>>  > Need a little help. 
Just set up an [EMAIL PROTECTED] box with 5 
Polycom>  > 501 phones. 
Everything works great except the messages button 
which>  > when pressed results 
in asterisk responding "Person at extension 
>  > 102 is on the phone. Please 
leave a message after the tone. I 
have>  > searched the web and 
several of the the asterisk mailing 
list>  > archive pages - but I 
haven't had any luck. Anyone have a suggestion? 
>  > 
___>  > 
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[Asterisk-Users] Automatic testing of my DIDs?

[Robert Augustyn]

Hi,
I wonder if someone wrote a script to test accessibility of DIDs.
This seems like a requirement to me giving the volatility of
providers/networks.
Thanks in advance.
robert

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RE: [Re] Re: [Asterisk-Users] Echo canceller on TE406 & Asterisk

[Robert Augustyn]

Darren,
Can you elaborate on what echocan did you use and how?
Thanks.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Darren Wright
> Sent: Friday, October 28, 2005 7:35 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: RE: [Re] Re: [Asterisk-Users] Echo canceller on 
> TE406 & Asterisk
> 
> I have given up totally on Digium based echo cancel, hardware 
> or software.  The KB1 is the best so far, but still 
> unacceptable.  I installed a hardware echocan FACING the T1 
> card in the asterisk box, and
> all is perfect.   No complaints from any of my clients since 
> taking that
> leap.
> 
> -Darren
> 
> 
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RE: [Asterisk-Users] Where can I find Polycom 600 config files?

[Robert Augustyn]

Thanks all for help.
I finally downloaded 1.6.2 version sip and everything seems to be working
fine.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Kristian Kielhofner
> Sent: Monday, October 17, 2005 11:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Where can I find Polycom 600 
> config files?
> 
> Robert Augustyn wrote:
> > Yep,
> > I tired these too ...
> > After the reboot it freezes.
> > The phone work ok with sip 1.4.2 though and previous 
> versions of these 
> > configuration files.
> > robert
> 
> Robert,
> 
>   If you are using pre 1.5.2 firmware, you need to use 
> the files in "/asterisk/pcom/", if you are using 1.5.2 you 
> need to use the files in "/asterisk/pcom/152/".
> 
> --
> Kristian Kielhofner
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RE: [Asterisk-Users] Where can I find Polycom 600 config files?

[Robert Augustyn]

Yep,
I tired these too ...
After the reboot it freezes.
The phone work ok with sip 1.4.2 though and previous versions of these
configuration files.
robert

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anthony Rodgers
> Sent: Monday, October 17, 2005 7:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Where can I find Polycom 600 
> config files?
> 
> You should see a list of files and folders - click on the 152 
> folder, and the sample files should be listed 
> <http://www.krisk.org/asterisk/ pcom/152/>.
> 
> What are you seeing?
> 
> Regards,
> --
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
> 
> 
> On 17-Oct-05, at 4:07 PM, Robert Augustyn wrote:
> 
> > Anthony,
> > If you referring to:
> > http://www.krisk.org/asterisk/pcom/
> > Then it does not seem to work for me 
> > robert
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > Anthony Rodgers
> > > Sent: Monday, October 17, 2005 5:57 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Where can I find Polycom 600
> > > config files?
> > >
> > > There is a link to excellent samples on the Polycom Phones
> > > page on voip-info.org.
> > >
> > > Regards,
> > > --
> > > Anthony Rodgers
> > > Business Systems Analyst
> > > District of North Vancouver
> > > Web: http://www.dnv.org
> > > RSS Feed: http://www.dnv.org/rss.asp
> > >
> > >
> > > On 17-Oct-05, at 1:50 PM, Robert Augustyn wrote:
> > >
> > > > Hi,
> > > > I am trying to configure my 600 phone using ftp.
> > > > The phone loaded the boot rom 2.6.2 and sip 1.5.2 but I 
> am having
> > > > problems
> > > > getting sip.cfg and phone1.cfg configuring 
> > > > Can someone send me examples of these two files, I 
> would very much
> > > > appreciate that.
> > > > robert
> > > >
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RE: [Asterisk-Users] Where can I find Polycom 600 config files?

[Robert Augustyn]

Anthony,
If you referring to:
http://www.krisk.org/asterisk/pcom/ 
Then it does not seem to work for me 
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anthony Rodgers
> Sent: Monday, October 17, 2005 5:57 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Where can I find Polycom 600 
> config files?
> 
> There is a link to excellent samples on the Polycom Phones 
> page on voip-info.org.
> 
> Regards,
> --
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
> 
> 
> On 17-Oct-05, at 1:50 PM, Robert Augustyn wrote:
> 
> > Hi,
> > I am trying to configure my 600 phone using ftp.
> > The phone loaded the boot rom 2.6.2 and sip 1.5.2 but I am having  
> > problems
> > getting sip.cfg and phone1.cfg configuring 
> > Can someone send me examples of these two files, I would very much
> > appreciate that.
> > robert
> >
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[Asterisk-Users] Where can I find Polycom 600 config files?

[Robert Augustyn]

Hi,
I am trying to configure my 600 phone using ftp.
The phone loaded the boot rom 2.6.2 and sip 1.5.2 but I am having problems
getting sip.cfg and phone1.cfg configuring 
Can someone send me examples of these two files, I would very much
appreciate that.
robert

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[Asterisk-Users] Can you use Polycom 500 with PoE Switch?

[Robert Augustyn]

Hi,
Do you need a special cable and power for that unit or you can run it of PoE
Switch?
Thanks.
robert

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[Asterisk-Users] What should I get for SOHO TDM card or sipura3000?

[Robert Augustyn]

Hi,
I am considering these two devices and would like to get your opionion on
which one to choose for the best voice quality.
I undestand that for two lines I would need two sipuras and one tdm with two
modules.
One limitation could be my asterisk box which is only 333 P2 with 512Ram.
Thanks in advance.
Robert


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[Asterisk-Users] Caller*ID failed checksum?

[Robert Augustyn]

Hi,
I am running ahh 1.0 and do not get caller id passed through to my phones.
In logs I can see following message:


May 13 16:57:53 VERBOSE[1376]: -- Starting simple switch on 'Zap/1-1'
May 13 16:57:57 NOTICE[1376]: Caller*ID failed checksum
May 13 16:57:59 NOTICE[1376]: Got event 2 (Ring/Answered)...


Any ideas what would be the reason?
I have searched for the answer but could not come up with anything ...
Thanks
robert


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RE: [Asterisk-Users] Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?

[Robert Augustyn]

Just get Dell SC420 and be done with it.
You should be able to get it for around 300 shipped with 2.8G cpu and 80G
HD.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Chuck Bunn
> Sent: Tuesday, April 12, 2005 9:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Low cost box for hosting 
> Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?
> 
> Hi,
> 
> Actually I guess what I am looking for is semi-sealed box 
> that I can add
> 1 or 2 PCI cards too. A regular PC work work in most cases 
> since I do not want a keyboard or mouse attached to it. I do 
> not want users screwing with the system. If it is sealed with 
> no monitor/keyboard/mouse then they can't screw it up very 
> easily. I guess I am looking for something that is somewhere 
> in between a PC and Linksys router box. One possibility might 
> be a thin client box, but I haven't found any sources for an 
> OEM box. I looked at the HP
> (http://h18004.www1.hp.com/products/thinclients/index_t5000.ht
> ml) thin clients but I can get a Dell Box for the same price 
> that does more.
> 
> Thanks
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RE: [Asterisk-Users] Asterisk@Home, gssftp and polycom report

[Robert Augustyn]

I second that.
Thanks for the good work.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Don Murray
> Sent: Tuesday, March 29, 2005 8:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] [EMAIL PROTECTED], gssftp and polycom report
> 
> 
> Recommendation for [EMAIL PROTECTED] : install another ftp option 
> rather than gssftp and tftp, such as vsftpd.
> 
> Details for anyone who is interested:
> 
> I posted a week or so ago about how we had 3 polycom phones 
> that were not updating their bootroms via the ftp server.  I 
> was using gssftp rather than tftp as it is recommended not to 
> use tftp with polycom phones on the wiki.  Gssftp is the 
> alternative supplied on [EMAIL PROTECTED]
> 
> Anyway, I got kerberos authentication errors with gssftp and 
> could not connect when I ftp'd by hand.  I fixed this by 
> changing the line:
> server_args = -l -a
> to
> server_args = -l
> in the /etc/xinetd.d/gssftp file.
> 
> I still got kerberos errors when ftping in by hand, but I 
> could put and get files no problem, there was only the errors 
> on log in.  The phones were able to connect and upload their 
> log files so I assumed that ftp access was working.
> 
> Anyway, after wasting a lot of time trying to get support 
> through Polycom, and playing around with things, I finally 
> decided that maybe these error messages were causing a 
> problem for the phones.
> 
> I googled around on these errors and "gssftp" and I did not 
> find much joy.  I couldn't find a definitive list of what 
> possibilities go into "server_args" and kerberos seems rather 
> arcane even for someone who has been a Linux user for 12 
> years like me.  (I guess I just moved from the "unsecure" 
> days of Linux directly to the "ssh" days of Linux and 
> bypassed the kerberos era.)
> 
> Anyway, the one thing I found by googling was that a lot of 
> people have the same problem I have.  I could not find anyone 
> saying "to fix that, do this configuration to gssftp".  
> Instead I found a lot of people saying "to fix that, install 
> a nicer ftp like vsftpd".  So, I did.  And the phones worked 
> immediately... just reboot and they could get their 
> configurations and bootrom updates and I was off  to the races.
> 
> So... mucking around with gssftp wasted a lot of my time and 
> now I really really hate it :)  My asterisk box is sitting 
> pretty behind a firewall so I don't have a lot of need for 
> network security.  With vsftpd I still get a funny kerberos 
> message (I'm wondering if I can just uninstall kerberos 
> because I really don't want to have anything to do with it... 
> any comments?)  Here is what I get when I ftp to my asterisk 
> box from the commandline:
> 
> 220 (vsFTPd 1.2.1)
> 530 Please login with USER and PASS.
> 530 Please login with USER and PASS.
> KERBEROS_V4 rejected as an authentication type
> 
> Anyway, I just thought this would be a good place to throw 
> the idea out of using a different ftp option for Asterisk.  
> There is an vsftpd RPM in the CentOS distribution.  Also, 
> thought I would report on this so that if anyone has the same 
> errors they might have a chance of finding this post and  
> following the same steps.
> 
> Don
> 
> 
> 
> 
> 
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RE: [Asterisk-Users] Asterisk@Home 0.7 released

[Robert Augustyn]

Has it been updated for AMP 1-10-007a?
Or manual update is required?
Thanks
Robert
Btw: great work!! 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Monday, March 28, 2005 8:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released
> 
> We had added a lot to this release to our one button install 
> of Asterisk. Now you can have even more features 
> automatically installed and configured.
> 
> Asterisk 1.0.7
> AMP 1-10-007
> Flash Operator Panel 0.20
> Redesigned WebMeetme
> weather agi scripts
> Midnight Commander
> 
> We have added some of our most requested features.
> 
> - Web Meetme is now installed by default and the
> meetme2 application is no longer needed.
> - we now have ZAP extension thanks to AMP 007
> - weather.agi reads the current weather report using text to speech
> 
> 
> 
>   
> __
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RE: [Asterisk-Users] VoIPJet and g.711

[Robert Augustyn]

Thanks,
Are you doing it by setting the lowest cost?
Is there anything in Asterisk which does it?
Thanks,
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Wojciech Tryc
> Sent: Sunday, March 13, 2005 12:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] VoIPJet and g.711
> 
> Robert,
> Nufone, but it all depends on the destination.
> For some is gafachi, for some is VoicePulse etc..
> W
> - Original Message -
> From: "Robert Augustyn" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> ; "'Justin Richards'" 
> <[EMAIL PROTECTED]>
> Sent: Sunday, March 13, 2005 12:09 PM
> Subject: RE: [Asterisk-Users] VoIPJet and g.711
> 
> 
> > Wojtek,
> > What are you using for your primary route?
> > robert
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
> >> [mailto:[EMAIL PROTECTED] On Behalf Of
> >> Wojciech Tryc
> >> Sent: Sunday, March 13, 2005 9:31 AM
> >> To: Justin Richards; Asterisk Users Mailing List -
> >> Non-Commercial Discussion
> >> Subject: Re: [Asterisk-Users] VoIPJet and g.711
> >>
> >> I can see errors on the console, g.729 and ilbc works no problem.
> >> I endup moving VoIPjet to the secondary route.
> >> Wojtek
> >> - Original Message -
> >> From: "Justin Richards" <[EMAIL PROTECTED]>
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> 
> >> Sent: Saturday, March 12, 2005 11:00 PM
> >> Subject: Re: [Asterisk-Users] VoIPJet and g.711
> >>
> >>
> >> >I am having problem with voipjet and g.711 (ulaw) as 
> well.  I tried
> >> > ilbc with no luck.  basically my outbound call connects, 
> i can hear
> >> > them talk, but they can't hear me.
> >> >
> >> > i am not getting errors in console with either ulaw or 
> ilbc, just no
> >> > audio to the called party.
> >> >
> >> > it worked great yesterday, and I haven't changed anything..  my
> >> > connection to voicepulse (same settings ad voipjet) works great.
> >> >
> >> > On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc
> >> <[EMAIL PROTECTED]> wrote:
> >> >> I am experiencing problems connecting to VoIPjet with
> >> g.711. It works
> >> >> with
> >> >> g.729 and ilbc. It used to work...
> >> >> Anyone?
> >> >> Regards,
> >> >> Wojtek
> >> >>
> >> >> ___
> >> >> Asterisk-Users mailing list
> >> >> Asterisk-Users@lists.digium.com
> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >> To UNSUBSCRIBE or update options visit:
> >> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >>
> >> > ___
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> >> > Asterisk-Users@lists.digium.com
> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >> > To UNSUBSCRIBE or update options visit:
> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >
> >>
> >> ___
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > ___
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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RE: [Asterisk-Users] VoIPJet and g.711

[Robert Augustyn]

Wojtek,
What are you using for your primary route?
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Wojciech Tryc
> Sent: Sunday, March 13, 2005 9:31 AM
> To: Justin Richards; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] VoIPJet and g.711
> 
> I can see errors on the console, g.729 and ilbc works no problem.
> I endup moving VoIPjet to the secondary route.
> Wojtek
> - Original Message -
> From: "Justin Richards" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Saturday, March 12, 2005 11:00 PM
> Subject: Re: [Asterisk-Users] VoIPJet and g.711
> 
> 
> >I am having problem with voipjet and g.711 (ulaw) as well.  I tried
> > ilbc with no luck.  basically my outbound call connects, i can hear
> > them talk, but they can't hear me.
> >
> > i am not getting errors in console with either ulaw or ilbc, just no
> > audio to the called party.
> >
> > it worked great yesterday, and I haven't changed anything..  my
> > connection to voicepulse (same settings ad voipjet) works great.
> >
> > On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc 
> <[EMAIL PROTECTED]> wrote:
> >> I am experiencing problems connecting to VoIPjet with 
> g.711. It works 
> >> with
> >> g.729 and ilbc. It used to work...
> >> Anyone?
> >> Regards,
> >> Wojtek
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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[Asterisk-Users] Polycom ip600 - how to eliminate echo?

[Robert Augustyn]

Hi,
I have noticable echo when I call on that device out using voipjet
termination.
Any idea how to eliminate that?
robert


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RE: [Asterisk-Users] VoIPJet

[Robert Augustyn]

How is your connection I am still getting all channels are busy 
thanks 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Peter Bowyer
> Sent: Wednesday, March 09, 2005 5:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] VoIPJet
> 
> On Wed, 9 Mar 2005 14:48:50 -0700, Wiley Siler 
> <[EMAIL PROTECTED]> wrote:
> > You are correct.  Apologies.
> 
> OK I guess. It would have taken way less time to re-check the 
> site in case you'd missed something than it took to flame me 
> for trying to help you...
> 
> Anyhow, the message has gone now, so I guess they've fixed it. 
> 
> Peter
> 
> --
> Peter Bowyer
> Email: [EMAIL PROTECTED]
> Tel: +44 1296 768003
> VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Is anybody having problems with sixtel?

[Robert Augustyn]

Hi,
My termination with sixtel stopped working, is it something I did or anybody
else is having the same problem.
I am attaching log:
*CLI>
-- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack
-- Goto (macro-dialout-default,s,4)
-- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack
-- Goto (macro-dialout-default,s,6)
-- Executing Dial("SIP/300-fbe0", "IAX2/sixTel/15197341953") in new
stack
-- Called sixTel/15197341953
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 00013ms  SCall: 1  DCall: 0 [205.234.133.203:4569]
   VERSION : 2
   CALLED NUMBER   : 15197341953
   CALLING NUMBER  : 300
   CALLING NAME: Robert
   LANGUAGE: en
   USERNAME: XX
   FORMAT  : 4
   CAPABILITY  : xx
   ADSICPE : 2
   DATE TIME   : 174436582

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 5ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
   AUTHMETHODS : 3
   CHALLENGE   : x
   USERNAME: x

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00119ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
   MD5 RESULT  : 38e3448d2773f728ca0d5249c5c651bf

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
   Timestamp: 00031ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
   FORMAT  : 4

-- Call accepted by 205.234.133.203 (format ulaw)
-- Format for call is ulaw
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00031ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ
   Timestamp: 10016ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP
   Timestamp: 10016ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK
   Timestamp: 10016ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ
   Timestamp: 10005ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP
   Timestamp: 10005ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK
   Timestamp: 10005ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: PING
   Timestamp: 20015ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: LAGRQ
   Timestamp: 20018ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: PONG
   Timestamp: 20015ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
   Unknown IE 046  : Present
   Unknown IE 048  : Present
   Unknown IE 049  : Present
   Unknown IE 050  : Present

Ignoring unknown information element 'Unknown IE' (46) of length 4
Ignoring unknown information element 'Unknown IE' (48) of length 4
Ignoring unknown information element 'Unknown IE' (49) of length 2
Ignoring unknown information element 'Unknown IE' (50) of length 4
Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 005 Type: IAX Subclass: ACK
   Timestamp: 20015ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 006 Type: IAX Subclass: LAGRP
   Timestamp: 20018ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[-01] -- OSeqno: 006 ISeqno: 006 Type: IAX Subclass: ACK
   Timestamp: 20018ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 006 Type: IAX Subclass: PING
   Timestamp: 20006ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: IAX Subclass: PONG
   Timestamp: 20006ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 006 Type: IAX Subclass: LAGRQ
   Timestamp: 20009ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Tx-Frame Retry[000] -- OSeqno: 007 ISeqno: 008 Type: IAX Subclass: LAGRP
   Timestamp: 20009ms  SCall: 1  DCall: 00056 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 007 Type: IAX Subclass: ACK
   Timestamp: 20006ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
Rx-Frame Retry[ No] -- OSeqno: 008 ISeqno: 008 Type: IAX Subclass: ACK
   Timestamp: 20009ms  SCall: 00056  DCall: 1 [205.234.133.203:4569]
-- Hungup 'IAX2/sixTel/1'
  == Spawn extension (macro-dialout-default, s, 6) exited non-zero on
'SIP/300-fbe0' in macro 'dialout-default'
  == Spawn extension (from-internal, 15197341953, 1) exited non-zero on
'SIP/

RE: [Asterisk-Users] Polycom Soundpoint 500/600 MiniBrowser

[Robert Augustyn]

Any chances of sharing it with us?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris HARIGA
Sent: Wednesday, March 02, 2005 6:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Polycom Soundpoint 500/600 MiniBrowser

Hi,

We use Polycom IP600 and the xml browser is working fine. We develop
in-house scripts to show on xml browser the conf rooms, parking, zap line
status, queue, stock exchange, etc.

Best regards,

Chris HARIGA


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Wednesday, March 02, 2005 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom Soundpoint 500/600 MiniBrowser

I'm trying to develop a company phone list accessible via the minibrowser
feature on the phone.

The pertinent section of ipmid.cfg is as follows:

 
  http://server/polycom/index.html";
mb.idleDisplay.refresh="300"/>
  http://server/polycom/index.html"/>
  
   

I can access the page via a regular browser, so I know it's working.
I'm running bootrom 2.6.1 and SIP 1.4.1.

When I press the "Services" button on the phone, it is dead.

Has anyone gotten this to work?

Thanks,
Adam

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[Asterisk-Users] Is using Sipura 2100 as SOHO main router good solution?

[Robert Augustyn]

 

Hi,
I got this device and it looks like I can get this to work as a router
connecting my local net to my DSL provider with QoS in place of my old
router.


I am thinking of using it as follows:
Dsl -> 2100 -> switch with * box and voip phones attached -> old router ->
other pcs


Does it sound like good plan? 
Thanks
robert


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RE: [Asterisk-Users] A hypothetical question...

[Robert Augustyn]

Marco,
Would you care to share more details about your 
configuration and user base?
It sounds like really nice solution.
robert


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Marco 
CastilloSent: Tuesday, February 15, 2005 7:34 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] A hypothetical question...

The 
complete configuration of such a system requires a lot more of information that 
the one you gave.But, at a glance,  Asterisk + SER is a good choice for 
this kind of venture. Asterisk can serve as the PSTN gateway (ISDN PRI 
connections primarily) and Voicemail server. SER can manage the billing and the 
VOIP-client part.
You 
can mount as many as Asterisk and SER servers as much as your traffic will 
require. So, you don't have to spend a lot of $$$ to mount such a large 
implementation. As I mention earlier, this is just a fast glimpse to a complete 
solution, I personally have such an implementation, and let me tell you that 
this works really great!!!
Hope 
this helps
 
Marco

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Rod 
  BaconSent: Tuesday, February 15, 2005 4:30 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] A 
  hypothetical question...
  
  I know this is casting a wide net, 
  but If you were charged with building a large, public VOIP network with 
  multiple PSTN gateways, the capacity to carry a lot of traffic and bill 
  clients accurately, what pieces (brands, makes, models) would you use to 
  assemble the solution? Assume that $$$ is not an 
  issue.
   
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RE: [Asterisk-Users] Uptime/reliability with SER, Asterisk

[Robert Augustyn]

How do you implement failover?
Thanks
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Blair
Sent: Tuesday, February 15, 2005 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Uptime/reliability with SER, Asterisk


Matthew:

  2 SER to 2 Asterisk (primary and secondary for each application) with
3 PSTN gateway. The boxes are in failover mode only. Each SER can fail over
to each PSTN gateway for outbound dialing, each gateway and phones can
failover to either SER box and either SER box can failover to either
Asterisk box. No load sharing.

_Steve

Matthew Boehm wrote:

>Steve,
> Do you have 1 SER to Many Asterisk? Or is it 1 to 1? If 1 to many, do 
>you do agents and queues? If so, how do you handle that across multiple *
boxes?
>
>THanks,
>Matthew
>
>- Original Message -
>From: "Steve Blair" <[EMAIL PROTECTED]>
>To: "Dana Olson" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - 
>Non-Commercial Discussion" 
>Sent: Monday, February 14, 2005 12:27 PM
>Subject: Re: [Asterisk-Users] Uptime/reliability with SER, Asterisk
>
>
>  
>
>>Our SER/Asterisk implementation is extremely stable if you define 
>>stable as the ability to deliver a set of features without either 
>>application crashing. We are a production environment with 75 users 
>>total. Asterisk is only used for voicemail. The only issue we have is 
>>that the audio (greeting or message) being play from Asterisk 
>>sometimes has a robotic or "stuttering" quality to it. I suspect this 
>>is latency in the data network but I have yet to figure it out.
>>
>>-Steve
>>
>>Dana Olson wrote:
>>
>>
>>
>>>Could anyone shed any light on how SER and/or Asterisk (stable 
>>>branch) has held up for them in that last while?
>>>
>>>Are you using SER and/or * in a production environment? Do you ever 
>>>restart the software or reboot the system? How many users are 
>>>utilizing the system? How many calls per day/concurrently?
>>>
>>>I read some uptimes and such on the mailing list from long ago, so I 
>>>was wondering what some more recent results were like. I'm running 
>>>Asterisk at home, but only since recently so my experience won't be a 
>>>good representation of the reliability and stability.
>>>
>>>Thanks in advance.
>>>___
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>>>
>>>  
>>>
>>--
>>
>>ISC Network Engineering
>>The University of Pennsylvania
>>3401 Walnut Street, Suite 221A
>>Philadelphia, PA 19104
>>
>>
>>voice: 215-573-8396
>>
>>   215-746-8001
>>
>>fax: 215-898-9348
>>
>>sip:[EMAIL PROTECTED]
>>
>>___
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RE: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED

[Robert Augustyn]

May I ask what you did?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent: Tuesday, February 15, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED

Its fixed and working great.
> 
> I was working on the dma issue that nenad pointed out and when i tried 
> to hdparm -d 1 /dev/hda my harddisk i got a permissioned denied error 
> (I was root) So i started researching a bit more and realised that 
> since i have a sata hdd and running it in IDE mode i cant start dma so 
> i just recompiled the kernel with sata drivers and scsi activated then 
> i change the bio SATA setting to SATA ENHANCED (nothing else seems to 
> work) and changed the root=/dev/hda2 to
> root=/dev/sda2 in the kernel boot options and i was on my way. now 
> when i tierd hdparm -d 1 /dev/sda i was told that scsi dosent have dma 
> so it didnt matter.
> 
> Next i started the wanpipe drivers and started asterisk i didnt get 
> any errors so to test th config i used
> 
> exten => 111,1,Dial(Zap/g1/301)
> exten => 111,2,Hangup
> 
> and dialed that extension 111 and the extension 301 rang. i had 
> started pri intense debugging and used show channels to make sure the 
> call went over the e1 channels.
> 
> i'm still testing it, it seems to be working great right now only 
> error i got was a FCS BAD or somthing like that once.
> 
> My motherboard is a SuperMicro P4SCI just for your information.

+++ Vikram Rangnekar [13/02/05 13:01 +0100]:
> 
> Does anyone have any experience ith configureing the sangoma A102 card 
> for testing using a e1 cross cable i've configured and installed the 
> cards properly even the lights on the card are green which proves that 
> my cross cable is properly built too. my problem is with asterisk 
> which gives me these errors
> 
> PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got 
> event: HDLC Bad FCS (8) on Primary D-channel of span 2 No D-channels 
> available! Using Primary on channel anyways 47!
> PRI: !! Not good - head of queue has not been transmitted yet
> 
> 
> I've tried everything i can think off with the wancfg configuration 
> files here is my zaptel and zapata configs.
> 
> span=1,0,0,ccs,hdb3
> bchan=1-15
> dchan=16
> bchan=17-31
> 
> span=2,1,0,ccs,hdb3
> bchan=32-46
> dchan=47
> bchan=48-62
> 
> --
> zapata.conf
> 
> switchtype=euroisdn
> signalling=pri_net
> group=1
> channel=>1-15
> channel=>17-31
> 
> group=2
> signalling=pri_cpe
> channel=>32-46
> channel=>48-62
> ---
> do i need to fool around with some jumpers on the card or something to 
> activate internal clock on the card. zttol says INTERNALLY CLOCKED for 
> both the ports. There are NO Alarms and no missed IRQ's I'm using 
> asterisk 1.0.5 on debian with 2.4.29 kernel
> 
> --
> regards
> Vikram (http://www.vicramresearch.com) 
> ___
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regards
Vikram (http://www.vicramresearch.com)
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RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.

[Robert Augustyn]

 I have problems getting into maintenance screen of AMP,
What is the user I should use? I must be missing something easy ...
Thanks
robert


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RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.

[Robert Augustyn]

I had the same problems,
I changed network card first, same problem then I changed the burner and
everything started to work.
Make sure that on the second pc you have different burner.
Oh and I use the nero 6 ... With no problems.
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger Hanson
Sent: Saturday, February 12, 2005 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup.

I tried downloading to a different PC - the download speed was much faster
(or was it the mirror I was using?) - 4MB/sec.  But again, I burned the
image and got the same error.

I suppose it's Nero that's messing it up?  I've never had any problems on
the many .iso's I've burned before with Nero.  I guess I could try a
different batch of CD-R's.

Is a network install possible for [EMAIL PROTECTED]

Someone send me a good CD and I'll paypal you a few bucks?

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

Sent: Saturday, February 12, 2005 4:50 PM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup.


> Roger Hanson wrote:
>> I've downloaded 2x and burned 2 cds and get an error "invalid 
>> compressed format (err=2) system halted" message both times.
>>
>> It'd be nice to have a MD5 to verify my download is OK.  It'd narrow 
>> down the problem to either the download or the burn, wouldn't it?
>>
>
> The other day I was getting problems with downloading files over 12mg 
> in size. They all were failing the checksum.
>
> Found out it was my driver for the nic card in my Linux box.  I was 
> using an RealTec. Changed the nic to an Intel and no problems after 
> that.
>
>>
>> - Original Message -
>> From: "Roderick A. Anderson" <[EMAIL PROTECTED]>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> 
>> Sent: Saturday, February 12, 2005 11:55 AM
>> Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on 
>> setup.
>>
>>
>>> Daniel Eboa wrote:
>>>
 I downloaded the iso file of the last release, but unable to burn 
 it on CD. Got error at 90%. Did anyone experience the same problem 
 ?
 Maybe the iso file is corrupted.


>>> Not as of approx 5:)) PM yesterday.  I downloaded, burned, and in 
>>> last stage of the install ( compiling * ) right now.
>>>
>>> I don't remeber if there was a md5sum for the iso, but a binary 
>>> error in hte download or bad hardware ( cd burner ) are the twom 
>>> main causes of this problem.
>>>
>>> Try another download.
>>>
>>>
>>> Rod
>>> --
>>
>> ___
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>
> 

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RE: [Asterisk-Users] asterisk@home basic

[Robert Augustyn]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Niemantsverdriet
Sent: Saturday, February 05, 2005 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] basic

I too had a hard time with email notfications. I knew that the * box needed
a smtp sever. I belive @home has send mail installed so you just need to add
a smart host. Sendmail.org has infomation on setting that all up.

Also try copying the correct /usr/share/zoneinfo file to /etc/localtime that
should clear up any problems.


On Sat, 5 Feb 2005 13:14:43 -, Steve Rawlings
<[EMAIL PROTECTED]> wrote:
> Apologies for asking something that must have been asked many times.  
> I'm running [EMAIL PROTECTED] v0.4 and can't get the * time to be local GMT.  
> Tried tzselect etc etc and added ntp server addresses to ntp.conf, * 
> still uses system time of EST so call logs are 5 hours behind.
> 
> Also, e-mail notifications of vm don't appear to be getting sent, I've 
> set voicemail.conf to include a valid e-mail address but I never receive
it.
> 
> Any pointers to subject posts would be appreciated, I'm not after 
> detailed replies just a pointer to where this might already be 
> documented, I've tried all the usual and obvious voip forums etc.
> 
> Steve
> 
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RE: [Asterisk-Users] Is Bell HDSL in Ontario good solution for VOIP?

[Robert Augustyn]

Sergey,
I am in Windsor ... That is 3.5 hours away ...
Btw: how much is colo there?
Robert
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Kuznetsov
Sent: Wednesday, February 02, 2005 7:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is Bell HDSL in Ontario good solution for
VOIP?

Robert,

Honestly, it's better to get colo at 151 Front St. from any big ISP company.


Robert Augustyn wrote:

>Hi,
>Have you tried it?
>Any comments would be greatly appreciated.
>I can have it at C$200, is that a good price?
>Thanks a lot.
>robert
>
>
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>  
>


--
All the Best!
Sergey.
=
Sergey Kuznetsov
President/CEO
  High Intellectual Technologies, Inc.

Web: http://www.hitcalls.com
 E-mail: [EMAIL PROTECTED]
Business phone: (416) 548-9700
   Mobile phone: (647) 287-8448
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[Asterisk-Users] Asterisk@home - problem getting console output ...

[Robert Augustyn]

Hi, 
I am connecting to the asterisk using asterisk -r command but I never get
anything on the  console?
How can I enable it?
Robert
Btw: it is version 0.4


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[Asterisk-Users] Is Bell HDSL in Ontario good solution for VOIP?

[Robert Augustyn]

Hi,
Have you tried it?
Any comments would be greatly appreciated.
I can have it at C$200, is that a good price?
Thanks a lot.
robert


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[Asterisk-Users] Asterisk@home problem installing CentOS ..

[Robert Augustyn]

Hi,
After I boot of the CD ( version 0.4 ) I get following error message:
"CD not found
The centos release 3.3 DC was not found in any of your cdroms drives.
Please insert the CentOS release 3.3 cd and press OK to retry."

Well I just boot of that cd so what is going on?
I have done some searches but can not find anything relevant.
Thanks.


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RE: [Asterisk-Users] FW: FAQ missing info? Asterisk@home V 0.4

[Robert Augustyn]

thanks 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff R
Glassman
Sent: Friday, January 28, 2005 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4


version 0.4 - 1/27/05
Asterisk 1.0.5
ztdummy
Integrated WebMeetMe
Bug fixes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert Augustyn
Sent: Friday, January 28, 2005 4:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4


What is new in .4?
Thanks
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff R
Glassman
Sent: Friday, January 28, 2005 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4

found it thanks

Was AMP updated?

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven Frazier
Sent: Friday, January 28, 2005 2:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4


It was released yesterday, I believe.


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jeff R 
> Glassman
> Sent: Friday, January 28, 2005 2:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] FW: FAQ missing info? 
> [EMAIL PROTECTED] V 0.4
> 
> 
> I had to redue the zapata.conf
> 
> Commented it out ans added,  Also changed default Zap g0 to Zap 1 
> (deleted Zap g0)
> 
> Where did you find [EMAIL PROTECTED] .4 all I see is 0.3
> 
> Jeff
> 
> [channels]
> language=en
> ;context=inbound-analog
> ;context=default
> context=from-pstn
> signalling=fxs_ks
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=yes
> channel => 1
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of dean 
> collins
> Sent: Friday, January 28, 2005 1:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4
> 
> 
> Just installed V 0.4 of [EMAIL PROTECTED]
> 
> Programmed up 3 sip budgetone extensions, they call call each other 
> fine.
> 
> Tried to dial '9' for an outside line through an X100P to a
> packet8 ATA but got 'all circuits are busy now'.
> 
> Here is the console output.
> 
> == Spawn extension (from-internal, h, 1) exited non-zero on 
> 'SIP/30-8d25'
> -- Executing SetGroup("SIP/30-5dde", "30") in new stack
> -- Executing Dial("SIP/30-5dde", "ZAP/g0/19172073420||") in new 
> stack
>   == Everyone is busy/congested at this time
> -- Executing Macro("SIP/30-5dde", "outisbusy") in new stack
> -- Executing Playback("SIP/30-5dde",
> "allison7/all-circuits-busy-now") in ne w stack
> -- Playing 'allison7/all-circuits-busy-now' (language 'en')
> -- Executing Playback("SIP/30-5dde",
> "allison7/pls-try-call-later") in new s tack
> -- Playing 'allison7/pls-try-call-later' (language 'en')
> -- Executing Macro("SIP/30-5dde", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/30-5dde", "w") in new stack
> -- Executing NoCDR("SIP/30-5dde", "") in new stack
> -- Executing Wait("SIP/30-5dde", "5") in new stack
>   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
> 'SIP/30-5dde' i n macro 'hangupcall'
>   == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 
> 'SIP/30-5dde' in  macro 'outisbusy'
>   == Spawn extension (from-internal, 919172073420, 103) exited 
> non-zero on 'SIP/ 30-5dde'
> -- Executing Macro("SIP/30-5dde", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/30-5dde", "w") in new stack
> -- Executing NoCDR("SIP/30-5dde", "") in new stack
> -- Executing Wait("SIP/30-5dde", "5") in new stack
>   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
> 'SIP/30-5dde' i n macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on 
> 'SIP/30-5dde' asterisk1*CLI>
> 
> 
> 
> any thoughts?
> 
> 
>

RE: [Asterisk-Users] FW: FAQ missing info? Asterisk@home V 0.4

[Robert Augustyn]

What is new in .4?
Thanks
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff R
Glassman
Sent: Friday, January 28, 2005 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4

found it thanks

Was AMP updated?

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven Frazier
Sent: Friday, January 28, 2005 2:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4


It was released yesterday, I believe.


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jeff R 
> Glassman
> Sent: Friday, January 28, 2005 2:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] FW: FAQ missing info? 
> [EMAIL PROTECTED] V 0.4
> 
> 
> I had to redue the zapata.conf
> 
> Commented it out ans added,  Also changed default Zap g0 to Zap 1 
> (deleted Zap g0)
> 
> Where did you find [EMAIL PROTECTED] .4 all I see is 0.3
> 
> Jeff
> 
> [channels]
> language=en
> ;context=inbound-analog
> ;context=default
> context=from-pstn
> signalling=fxs_ks
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=yes
> channel => 1
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of dean 
> collins
> Sent: Friday, January 28, 2005 1:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4
> 
> 
> Just installed V 0.4 of [EMAIL PROTECTED]
> 
> Programmed up 3 sip budgetone extensions, they call call each other 
> fine.
> 
> Tried to dial '9' for an outside line through an X100P to a
> packet8 ATA but got 'all circuits are busy now'.
> 
> Here is the console output.
> 
> == Spawn extension (from-internal, h, 1) exited non-zero on 
> 'SIP/30-8d25'
> -- Executing SetGroup("SIP/30-5dde", "30") in new stack
> -- Executing Dial("SIP/30-5dde", "ZAP/g0/19172073420||") in new 
> stack
>   == Everyone is busy/congested at this time
> -- Executing Macro("SIP/30-5dde", "outisbusy") in new stack
> -- Executing Playback("SIP/30-5dde",
> "allison7/all-circuits-busy-now") in ne w stack
> -- Playing 'allison7/all-circuits-busy-now' (language 'en')
> -- Executing Playback("SIP/30-5dde",
> "allison7/pls-try-call-later") in new s tack
> -- Playing 'allison7/pls-try-call-later' (language 'en')
> -- Executing Macro("SIP/30-5dde", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/30-5dde", "w") in new stack
> -- Executing NoCDR("SIP/30-5dde", "") in new stack
> -- Executing Wait("SIP/30-5dde", "5") in new stack
>   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
> 'SIP/30-5dde' i n macro 'hangupcall'
>   == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 
> 'SIP/30-5dde' in  macro 'outisbusy'
>   == Spawn extension (from-internal, 919172073420, 103) exited 
> non-zero on 'SIP/ 30-5dde'
> -- Executing Macro("SIP/30-5dde", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/30-5dde", "w") in new stack
> -- Executing NoCDR("SIP/30-5dde", "") in new stack
> -- Executing Wait("SIP/30-5dde", "5") in new stack
>   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
> 'SIP/30-5dde' i n macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on 
> 'SIP/30-5dde' asterisk1*CLI>
> 
> 
> 
> any thoughts?
> 
> 
> Cheers,
> Dean
> 
> 
> -Original Message-
> From: dean collins
> Sent: Friday, January 28, 2005 10:56 AM
> To: 'andrew'
> Subject: RE: FAQ missing info
> 
> Btw, do you need the pstn line for the X100P plugged in while 
> running the install?
> 
> Just downloaded and burning the cd now.
> 
> 
> 
> -Original Message-
> From: andrew [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 28, 2005 12:04 AM
> To: dean collins
> Subject: Re: FAQ missing info
> 
> What card do you have? Version 0.4 supports the  X100P automatically.
> 
> 
> --- dean collins <[EMAIL PROTECTED]>
> wrote:
> 
> > Message body follows:
> >
> > hi, I might be missing something basic but are you
> > supposed
> > to edit zaptel file or is it supposed to do it
> > automatically?
> >
> > I've posted this question on the asterisk and amp
> > lists but no
> > one seems to be able to answer me on this.
> >
> > Cheers,
> > [EMAIL PROTECTED]
> >
> > --
> > This message has been sent to you, a registered 
> SourceForge.net user,
> > by another site user, through the SourceForge.net
> > site.  This message
> > has been delivered to your SourceForge.net mail
> > alias.  You may reply
> > to this message using the "Reply" feature of your
> > email client, or
> > using the messaging facility of SourceForge.net at:
> >
> https://sourceforge.net/sendmessage.php?touser=1157926
> >
> 
> 

RE: [Asterisk-Users] Where can I find good doc on AGI?

[Robert Augustyn]

Thanks,
I have seen that but this is over 2 years old, does it mean that it is still
current?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vassil Kolarov
Sent: Friday, January 28, 2005 10:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Where can I find good doc on AGI?


http://home.cogeco.ca/~camstuff/agi.html



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Friday, January 28, 2005 4:28 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Where can I find good doc on AGI?

Hi,
I have searched the list/Wiki, web and I am not able to find a decent
documentation of the AGI/FastAGI interface with examples.
Am I looking in wrong places? 
Help will be greatly appreciated.
Robert





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[Asterisk-Users] Where can I find good doc on AGI?

[Robert Augustyn]

Hi,
I have searched the list/Wiki, web and I am not able to find a decent
documentation of the AGI/FastAGI interface with examples.
Am I looking in wrong places? 
Help will be greatly appreciated.
Robert





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RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCardApplicationforAsterisk

[Robert Augustyn]

 
Would you care to elaborate?
Robert

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat
Sent: Wednesday, January 26, 2005 1:36 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW
CallingCardApplicationforAsterisk

Once you compare Postgress and MySQL you will never want to go back to
MySQL.

-Original Message-----
From: Robert Augustyn [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 10:07 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard
ApplicationforAsterisk

NICE!
I understand that it works against Postgress, any idea what it would take to
port it to mysql if anything?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski
Sent: Wednesday, January 26, 2005 12:05 PM
To: Asterisk-Users Mailing-list
Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application
forAsterisk

Hello everyone,


If you want to know why I am so tired today :D Check this CallingCard
Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night!


Briefly, AreskiCC is an AGI script and PHP-Web application which greatly
handle the complete CallingCard System.


FEATURES - AGI :
  * Authenticate with the use of a Cardnumber 
the Cardnumber can also be defined as accountcode into sip.conf,
iax.conf, etc.. 
  * take care of multiple calls using the same Cardnumber 
  * Caller gets informed about his credit 
Announce the remaining credit
  * Caller is requested to enter a destination number 
  * Announce the maximal call time for the given destination number 
It calculates the remaining duration of the actual call (based
on tariffrate tables), informs the caller about this and sets a
timeout
  * Interupt the call if the card balance gets zero 
Warn the caller about the call interupt 60 & 30 seconds before
the call gets interupted
  * It connects the Caller to the destination through the configured
trunk 
note : different trunks can be configured and associated by
prefix
  * After disconnecting the call AGI updates the credit and stores
the concerning Call-Detail-Records with CallingPartyNumber,
CalledPartyNumber, CallSetupTime, Duration, Charge and the
remaining credit


FEATURES - WEB INTERFACE:
  * CARD/CUSTOMERS
  * List customers
  * Refill customer
  * CARD/CUSTOMERS
  * List customers/cards
  * Refill customer/card
  * Create customer/card
  * Generate customers/cards
  * BILLING
  * View money situation
  * View Payment
  * Add new Payment
  * RATECARD
  * List Tariffplan
  * Create new Tariffplan
  * Define Tariffplan
  * TRUNK
  * List Trunk
  * Add Trunk
  * CALL REPORT - BALANCE 

Last note : It's distributed under GNU GPL Licence.



I hope there will have a big interest for the soft, I am waiting your
feedbacks... 

Regards,
/Areski





-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_

Belaïd Arezqui
www.areski.net
E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com
 

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RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

[Robert Augustyn]

Sweet !

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Wednesday, January 26, 2005 1:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

The 1.4.1 firmware and the 2.6.1 bootrom are also now on
http://www.freedomphones.net/polycom/files/

MATT---

-Original Message-
From: mattf
Sent: Wednesday, January 26, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1


the 1.3.4 firmware is available on
http://www.freedomphones.net/polycom/files/


MATT---


-Original Message-
From: Chad Scott [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1


Noah, I could really use 1.3.4... however, a better question might be 
how are *you* getting 1.3.4?  I can't seem to get this from anyone, 
including my reseller.

On Jan 26, 2005, at 8:42 AM, Noah Miller wrote:

> Hi Chris -
>
>> I am getting to my wits end with these phones (and so is my boss).  I 
>> am
>> getting an random echo on these phones and I have an issue opened with
>> Polycom and its been in their research and development department for
>> almost a month with no results.
> I'm amazed it got that far with Polycom.  My experience is that they 
> do not support these phones at all.
>
>> I have noticed that I get a message "RFC3389 support incomplete.  Turn
>> off on client if possible" in asterisk. I have researched this and 
>> made
>> the change in ipmid.cfg (see below), but I am still getting this RFC
>> error.
>>
>> --- ipmid.cfg 
>> 
>>> qos.ip.rtp.max_throughput="0" qos.ip.rtp.max_reliability="0"
>> qos.ip.rtp.min_cost="0" qos.ip.rtp.precedence="0"/>
>>> tcpIpApp.port.rtp.filterByPort="0" tcpIpApp.port.rtp.forceSend=""
>> tcpIpApp.port.rtp.mediaPortRangeStart=""/>
>> - end 
>>
>> I am just wondering if anyone can help me troubleshoot the echo and 
>> RFC
>> error so I don't have to pull the entire phone system out and purchase
>> an entire new system.
> What version of * are you using?  I'm using 1.0.2 with Polycom 1.3.4 
> firmware on IP600's and I haven't seen any of these problems.  If 
> you'd like the 1.3.4 version of the firmware, just let me know off 
> list.
>
> Stupid Question:  Is the echo on all calls or just on calls to/from 
> the PSTN?  If just PSTN calls, do you have any echo cancellation 
> enabled?
>
> Thanks,
> Noah
>
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> !DSPAM:41f7d1bf129142063020194!

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RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

[Robert Augustyn]

Matt,
How can we get newer files stored on that site?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Wednesday, January 26, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

the 1.3.4 firmware is available on
http://www.freedomphones.net/polycom/files/


MATT---


-Original Message-
From: Chad Scott [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1


Noah, I could really use 1.3.4... however, a better question might be how
are *you* getting 1.3.4?  I can't seem to get this from anyone, including my
reseller.

On Jan 26, 2005, at 8:42 AM, Noah Miller wrote:

> Hi Chris -
>
>> I am getting to my wits end with these phones (and so is my boss).  I 
>> am getting an random echo on these phones and I have an issue opened 
>> with Polycom and its been in their research and development 
>> department for almost a month with no results.
> I'm amazed it got that far with Polycom.  My experience is that they 
> do not support these phones at all.
>
>> I have noticed that I get a message "RFC3389 support incomplete.  
>> Turn off on client if possible" in asterisk. I have researched this 
>> and made the change in ipmid.cfg (see below), but I am still getting 
>> this RFC error.
>>
>> --- ipmid.cfg 
>> 
>>> qos.ip.rtp.max_throughput="0" qos.ip.rtp.max_reliability="0"
>> qos.ip.rtp.min_cost="0" qos.ip.rtp.precedence="0"/>
>>> tcpIpApp.port.rtp.filterByPort="0" tcpIpApp.port.rtp.forceSend=""
>> tcpIpApp.port.rtp.mediaPortRangeStart=""/>
>> - end 
>>
>> I am just wondering if anyone can help me troubleshoot the echo and 
>> RFC error so I don't have to pull the entire phone system out and 
>> purchase an entire new system.
> What version of * are you using?  I'm using 1.0.2 with Polycom 1.3.4 
> firmware on IP600's and I haven't seen any of these problems.  If 
> you'd like the 1.3.4 version of the firmware, just let me know off 
> list.
>
> Stupid Question:  Is the echo on all calls or just on calls to/from 
> the PSTN?  If just PSTN calls, do you have any echo cancellation 
> enabled?
>
> Thanks,
> Noah
>
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RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk

[Robert Augustyn]

NICE!
I understand that it works against Postgress, any idea what it would take to
port it to mysql if anything?
robert 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski
Sent: Wednesday, January 26, 2005 12:05 PM
To: Asterisk-Users Mailing-list
Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application
forAsterisk

Hello everyone,


If you want to know why I am so tired today :D Check this CallingCard
Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night!


Briefly, AreskiCC is an AGI script and PHP-Web application which greatly
handle the complete CallingCard System.


FEATURES - AGI :
  * Authenticate with the use of a Cardnumber 
the Cardnumber can also be defined as accountcode into sip.conf,
iax.conf, etc.. 
  * take care of multiple calls using the same Cardnumber 
  * Caller gets informed about his credit 
Announce the remaining credit
  * Caller is requested to enter a destination number 
  * Announce the maximal call time for the given destination number 
It calculates the remaining duration of the actual call (based
on tariffrate tables), informs the caller about this and sets a
timeout
  * Interupt the call if the card balance gets zero 
Warn the caller about the call interupt 60 & 30 seconds before
the call gets interupted
  * It connects the Caller to the destination through the configured
trunk 
note : different trunks can be configured and associated by
prefix
  * After disconnecting the call AGI updates the credit and stores
the concerning Call-Detail-Records with CallingPartyNumber,
CalledPartyNumber, CallSetupTime, Duration, Charge and the
remaining credit


FEATURES - WEB INTERFACE:
  * CARD/CUSTOMERS
  * List customers
  * Refill customer
  * CARD/CUSTOMERS
  * List customers/cards
  * Refill customer/card
  * Create customer/card
  * Generate customers/cards
  * BILLING
  * View money situation
  * View Payment
  * Add new Payment
  * RATECARD
  * List Tariffplan
  * Create new Tariffplan
  * Define Tariffplan
  * TRUNK
  * List Trunk
  * Add Trunk
  * CALL REPORT - BALANCE 

Last note : It's distributed under GNU GPL Licence.



I hope there will have a big interest for the soft,
I am waiting your feedbacks... 

Regards, 
/Areski





-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_

Belaïd Arezqui
www.areski.net
E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com
 

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RE: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

[Robert Augustyn]

 Noah,
How can I contact you of the list?
robert

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, January 26, 2005 11:43 AM
To: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Polycom IP 600 - 1.3.1

Hi Chris -

>I am getting to my wits end with these phones (and so is my boss).  I 
>am getting an random echo on these phones and I have an issue opened 
>with Polycom and its been in their research and development department 
>for almost a month with no results.
>  
>
I'm amazed it got that far with Polycom.  My experience is that they do not
support these phones at all.

>I have noticed that I get a message "RFC3389 support incomplete.  Turn 
>off on client if possible" in asterisk. I have researched this and made 
>the change in ipmid.cfg (see below), but I am still getting this RFC 
>error.
>
>--- ipmid.cfg 
>
>qos.ip.rtp.max_throughput="0" qos.ip.rtp.max_reliability="0"
>qos.ip.rtp.min_cost="0" qos.ip.rtp.precedence="0"/>
>tcpIpApp.port.rtp.filterByPort="0" tcpIpApp.port.rtp.forceSend=""
>tcpIpApp.port.rtp.mediaPortRangeStart=""/>
>- end 
>
>I am just wondering if anyone can help me troubleshoot the echo and RFC 
>error so I don't have to pull the entire phone system out and purchase 
>an entire new system.
>  
>
What version of * are you using?  I'm using 1.0.2 with Polycom 1.3.4
firmware on IP600's and I haven't seen any of these problems.  If you'd like
the 1.3.4 version of the firmware, just let me know off list.

Stupid Question:  Is the echo on all calls or just on calls to/from the
PSTN?  If just PSTN calls, do you have any echo cancellation enabled?

Thanks,
Noah

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[Asterisk-Users] Perfect billing solution for *?

[Robert Augustyn]

Hi,
I wonder what would you consider a perfect billing solution for *?
What would it have to have and what would be nice 
robert


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RE: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600

[Robert Augustyn]

If you have it, can I get a copy please, or possibly can you send it to the
keeper of http://www.freedomphones.net/polycom/files/ 
I am looking for the latest boot image too.
Thanks.
robert

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Devenijn
Sent: Tuesday, January 25, 2005 5:30 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600

Does somebody have this new firmware from/for Polycom ?

Thanks 

Michael

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[Asterisk-Users] How to reset IP600 with no password?

[Robert Augustyn]

Hi,
I want to reset IP600 to the factory settings but when I press 468* and hold
it asks for password?
Is there another way?
Robert


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