[asterisk-users] POKE from command line

[Gary Carr]

Is it possible to issue the POKE to a end point from the CLI? Our 
asterisk servers is not seeing some end points drop off and I would like 
to create a script to manually check end points.



Thanks!


Gary


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Re: [asterisk-users] DIDForSale spam

[Gary Carr]

I received the same spam myself.


Regards,


Gary Carr

List users,

Did anyone else recently receive spam from DIDForSale with the subject
DIDForSale 2012 achievements?  I suspect that they are using this
list to harvest email addresses and think they should be called out on
this poor business practice if that is the case.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] call extension play sound file then connect caller

[Gary Carr]

-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Gary Carr
Sent: Wednesday, October 03, 2012 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call extension play sound file then connect 
caller


I am trying to setup a context to take a inbound call, hold the call, 
connect to
an external number, play a sound file to the external number, then 
connect

the inbound caller to the external number.

My thought was to accept the call and place them in a parking lot. Then 
call
the external number, play the sound file and connect the inbound caller 
to

the external number.


Is this even possible and if so, is this the best approach?


Thank you in advance.



You might look into FollowMe, especially if you want the external number 
to have a choice of whether or not to accept the call.


A very high level overview is here: 
http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ 
(though that gave me enough to get started)





Thanks for the reply. I tried using FollowMe as it seemed like the perfect 
solution, however I was unable to play the sound file then connect the 
caller. I would like to bypass the need to press the 1 to accept the call.



Thanks Again! 



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[asterisk-users] call extension play sound file then connect caller

[Gary Carr]

I am trying to setup a context to take a inbound call, hold the call, 
connect to an external number, play a sound file to the external number, 
then connect the inbound caller to the external number.


My thought was to accept the call and place them in a parking lot. Then call 
the external number, play the sound file and connect the inbound caller to 
the external number.



Is this even possible and if so, is this the best approach?


Thank you in advance.





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[asterisk-users] white noise on conference

[Gary Carr]

I am trying to track down a white noise problem we are having in our conference 
rooms. If there are 3 or 4 users in the conference the quality is good. After 
we get more users in the conference we develop a white noise that gets louder 
as more users come online. I have tried both meetme and confbridge. I am 
running 1.8.16.0 compiled from source.

Can anyone provide some insight on where to look or anything to tweak to 
resolve this?--
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[asterisk-users] confbridge command not found

[Gary Carr]

Currently running version 1.8.16.0 and trying to manage confbridge rooms and 
users. When I try to use the confbridge cli command I get a command not found 
error.


CLI confbridge
No such command 'confbridge' (type 'core show help confbridge' for other 
possible commands)


I've tried googling this but did not get anywhere. How can I enable the 
confbridge commands?


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Re: [Asterisk-Users] Static on TDM Zaptel FXO

[Gary Carr]

I have that exact setup and am getting a echo at the beginning of a inbound 
call. What gain settings work best for you?


Gary

Hello Bryce,
Gain settings do seem to have an effect.  I am going from a Cisco
7960AsteriskZap TDM CardPOTS
Thanks,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryce
Chidester
Sent: Wednesday, May 18, 2005 1:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO
That type of echo is usually caused by incorrectly (or not at all) tuned
gain settings in zapata.conf. I don't know what kind of phones you're
using, but for Asterisk to even be able to detect DTMF tones on our
Sayson / Aastra 390s and 480s, our FXS channels are set to -5.0 on both
rx and txgain. If you're using externally-powered phones (as in not your
ordinary joe-schmoe analog phone), I have found that they're usually
pretty hot (loud) and Asterisk can't understand what is said.
Good luck!
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On May 17, 2005, at 19:00, Gregory Wiktor - ADCom Corp. wrote:

On a recompile of the kernel I now get a 99.98 average.
Static is gone, although quality so far seems not quite there yet.
I am also experiencing an odd local echo.  I can hear a slight echo
locally, but the other end sounds fine, and the other end does not get

echo.
Even with the pots disconnected, you can hear it.  The static would be

on all calls.  Hooking up a normal phone was ok.  The sipsip phones
are perfect too, it was only happening on the zap channel...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Parr
Sent: Monday, May 16, 2005 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO
On 5/16/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:

Hello All,
I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy
static.
Even with the pots line disconnected, if I do a dial I still get

static.

This way I know it's not the line, but rather something on the card.
I tried alternate pci slots.
This card has a power connector, does anyone know what the power
requirements are?  The unit is in a small case with a 2.4ghz p-4 and
512mb ram, on an intel board with 533fsb.  All other functions are

fine.

I am using the latest CVS on Debian 2.6test
Anyone experience this?

Have you tried a different phone? Does the static appear immediately
when you pick up the phone? Or on the second or third time?
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[Asterisk-Users] card description when running pspci -vb

[Gary Carr]

What should the description be for a digium FXS card when running the 
lspci -b? Mine shows the following.


00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN 
interface
   Subsystem: Unknown device b100:0003
   Flags: bus master, medium devsel, latency 32, IRQ 10
   I/O ports at ec00 [size=256]
   Memory at d000 (32-bit, non-prefetchable) [size=4K]
   Capabilities: [40] Power Management version 2


Thanks,
Gary
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[Asterisk-Users] slight echo on incoming call

[Gary Carr]

Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am
hearing a brief echo on our Cisco 7960 phones when a incoming call is
answered. After a few seconds of conversation the echo disappears. There is
no echo on outbound calls or transferred calls. After a search of the
mailing list, wikki, and google I have tried the following to no avail.
echocanel=yes as well as 16, 32, 128, and 256
echotraining=yes as well as 800
echocancelwhenbridged=yes
I have also modified the rxgain= and txgain= settings.
Short of recompiling the zaptel with agreessive echotraining can anyone
suggest some other things I may be missing?

Thanks,
Gary

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[Asterisk-Users] brief echo on incoming call

[Gary Carr]

Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am 
hearing a brief echo on our Cisco 7960 phones when a incoming call is 
answered. After a few seconds of conversation the echo disappears. There is 
no echo on outbound calls or transferred calls. After a search of the 
mailing list, wikki, and google I have tried the following to no avail.

echocanel=yes as well as 16, 32, 128, and 256
echotraining=yes as well as 800
echocancelwhenbridged=yes
I have also modified the rxgain= and txgain= settings.
Short of recompiling the zaptel with agreessive echotraining can anyone 
suggest some other things I may be missing?


Thanks,
Gary

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Re: [Asterisk-Users] using * for Internet call waiting

[Gary Carr]

You need a V92 capable modem for your client and a V92 capable access
server for you.  The feature is called modem on hold, it lets you
pick up a call without loosing your internet connection, and resume
the dialup session after hangup. The only feature you need for your
telco is call waiting. It does not need forward on busy. Regards,

That's one way of doing it. The other is call forward busy and how most of 
the existing services do it.


Gary

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Re: [Asterisk-Users] using * for Internet call waiting

[Gary Carr]

The * box would sit in a CO connected via PRIs.

Gary

Gary Carr wrote:
Wondering if it is possible or if something already exist to setup * to 
offer Internet Call Waiting. For those that do not know what it is, it's 
a small application that runs on a users computer that will pop up a 
window letting them know they have a incoming call and who it is from 
then they can choose to take the call which will disconnect their dialup 
modem and ring their phone or send the call to voice mail.
That doesn't really make sense if the * box is in your house because
if the phone line is tied up for a dialup call, then the * box doesn't
have a phone line to receive the call either (unless you had call hunting
in which case you wouldn't need the feature in the first place).  This
sounds like the sort of feature that can only be offered on the
central office side which can know your line is tied up and then know
to email/alert you.
The other scenario is having an * box in a call center that is forwarding
calls to agents and notifies them by TCP/IP if when it tries their
extension and gets a busy signal.  This sounds possible, but I don't
think it's what you meant.
Steve
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Re: [Asterisk-Users] using * for Internet call waiting

[Gary Carr]

The TDM part is pretty simple. The end user needs the call forward busy 
feature on thier line with the calls being forwarded to the * server. Taking 
it from there and sending it to a app on the users machine is whats left. I 
was thinking it could be sent with sip and a long timeout value.


Gary

I've seen this service done with AOL, I was curious how it was done on
standard phone lines.  Was it something the coordinated with the telco
in some sort of hunt group configuration or something of that nature?
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.
Thanks,
Gary
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Re: [Asterisk-Users] using * for Internet call waiting

[Gary Carr]

That's pretty close to what we are looking for but we want the user to have 
the option of taking the call which would disconnect the modem connection 
and allow the call to ring thru to the phone. Not sure how to accomplish 
that. I am sure our programmer could code a client but he has no experience 
with *. If we can figure out that part we could come up with something.


Gary

I'm an ISP, what I would like is a client for the dialup customer to run.
They would use call fwd busy to my did on an asterisk box.
I'd signal and they could click on button (URL) to download .wav file in
asterisk voice mail.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mojo with
Horan  Company, LLC
Sent: Thursday, April 21, 2005 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] using * for Internet call waiting
I once tried the pagoo service.  Seems I had to ask the telco for Call
Forward Busy, and provide them with the toll free number pagoo gave me
for their service.  When the forwarded call is received by their
systems, they  would see _my_ callerid information, and thus know to
contact my computer for the notification purpose.
Also, not sure if this is on track with what you want, but I've used
jabber_client.pl tied into my dialplan to popup the callerid info of an
incoming call on my screen..  I could then choose to answer the call or
let it ring to voicemail.  Seems the jabber client Neos has
well-designed popups.
links:
http://jabberd.jabberstudio.org/2/
for the jabber_alert.pl script, allows sending jabber msgs from cmd line.
http://www.neosmt.com/
for a jabber client that pops up incoming messages. Note, this is also
an H.323 client.  Haven't tried it with * yet, but I have been meaning to.
Here's the specific Dialplan line I use:
[inpstn]
exten = s,2,TrySystem(echo Incoming call from :${CALLERID} |
jabber_alert.pl -e [EMAIL PROTECTED] -n [EMAIL PROTECTED] -w
senders_password)
Because it can sometimes take 2 or 3 seconds to send the jabber message
on my network, I use TrySystem instead of System, which blocks, waiting
for the return code from the command I passed.  Because the return code
is prolly irrelevant, you'd most likely want to use TrySystem too...
hope this helps :)
Moj

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.
Thanks,
Gary
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[Asterisk-Users] using * for Internet call waiting

[Gary Carr]

Wondering if it is possible or if something already exist to setup * to 
offer Internet Call Waiting. For those that do not know what it is, it's a 
small application that runs on a users computer that will pop up a window 
letting them know they have a incoming call and who it is from then they can 
choose to take the call which will disconnect their dialup modem and ring 
their phone or send the call to voice mail.

Thanks,
Gary
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Re: [Asterisk-Users] RE: Re: MGCP to Inter Tel system

[Gary Carr]

-this is very true, however, the current version of the Axxess software
(9.0) supports SIP trunking natively on the IPRC.  I just got my Axxess
upgraded and am salivating to get * connected to it.

Hmm, so 9.0 is out and it supports SIP natively. How did you plan to 
integrate the 2?


Gary
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Re: [Asterisk-Users] ATA that actually work with T.38

[Gary Carr]

No, the PAP2's are. The PAP2-NA is for any provider.

Gary
- Original Message - 
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, February 14, 2005 11:33 AM
Subject: Re: [Asterisk-Users] ATA that actually work with T.38


Hi Gary,
Aren't those all tied to service providers now?
Regards,
Steve
Gary Carr wrote:
We use the PAP-2NA with fax machines and have not had any problems.
Gary
Hi,
I am implementing T.38, and finding a problem getting boxes that work 
with T.38 for testing. A lot (maybe most) ATAs now claim to support 
T.38, but I'm finding a lot of these lie. I have one box here that just 
crashes when it hears a fax tone. :-)

I'm looking for boxes known to implement T.38 properly, and which really 
work in the real world.

Regards,
Steve

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Re: [Asterisk-Users] ATA that actually work with T.38

[Gary Carr]

That site is correct. You have to be authorized by Linksys to order the 
product from a distributor but they will work with any VoIP service. We use 
them with our * service.


Gary

Quoting Gary Carr [EMAIL PROTECTED]:
You might want to tell that to these guys:
http://www.voipsupply.com/product_info.php?products_id=317
regards,
Paul

No, the PAP2's are. The PAP2-NA is for any provider.

Gary
- Original Message - 
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 14, 2005 11:33 AM
Subject: Re: [Asterisk-Users] ATA that actually work with T.38

 Hi Gary,

 Aren't those all tied to service providers now?

 Regards,
 Steve


 Gary Carr wrote:

 We use the PAP-2NA with fax machines and have not had any problems.

 Gary

 Hi,

 I am implementing T.38, and finding a problem getting boxes that work
 with T.38 for testing. A lot (maybe most) ATAs now claim to support
 T.38, but I'm finding a lot of these lie. I have one box here that 
 just

 crashes when it hears a fax tone. :-)

 I'm looking for boxes known to implement T.38 properly, and which 
 really

 work in the real world.

 Regards,
 Steve


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Re: [Asterisk-Users] ANNOUNCEMENT:NEWCallingCardApplicationforAsterisk

[Gary Carr]

Anyone have a copy of the DB_areskicc.psql file mentioned in the AGI tar 
file for this new application?


Thanks,
Gary
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[Asterisk-Users] looking to hire for configuration of call routing based on RDNIS

[Gary Carr]

We need to implement incoming call routing based on RDNIS and I would like 
to contract with someone who can set this up. We are forwarding calls to * 
using call forward no answer and call forward busy on the customers line and 
the RDNIS information is suppose to passed to *. I need to route the call to 
the proper voice mailbox based on that information.

If you can configure this for us please email me directly.

Thanks,

Gary Carr
President/CEO
705A Wesley Pines Rd.
COL Networks, Inc.
Lumberton, NC 28358
Phone: 910-402-5011
Fax: 910-618-9027
Check us out at: www.carolina.net

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Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Gary Carr]

I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d, IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
   -- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read: Call 
rejected
by 200.53.121.233: No such context/extension
   -- Hungup 'IAX2/200.53.121.233:4569/4'
 == No one is available to answer at this time

Regards,
Gary
- Original Message - 
From: Miguel Cavazos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 10:13 AM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


any feedback would be awsome, the idea is to fill in the 30 channels of 
the E1 all at the same time and see how stable it can be

On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone there 
to
call but I can call some hotels
in the area for possible reservations and perhaps ticket information for 
the
theater.

- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
I will let anyone make FREE LOCAL calls to Mexico City till saturday or
maybe until monday to see how stable this can be with REAL traffic. Add
this to your extensions.conf only gsm as a codec is going to be
permitted.
exten =
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
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--
Saludos,
Miguel Cavazos
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Re: [Asterisk-Users] Re: R2/MFC Mexico FREE calls to test chan_unicall(Miguel Cavazos)

[Gary Carr]

I made a few test calls to the airport but don't speak spanish so I had no 
idea what they were saying :)

Anyone have Marco Antonio Barrera's phone number :-()

Gary

I just made three calls. stayed on the phone about 5 to 10 minutes per 
call
works great! Crystal Clear!!

Juan
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[Asterisk-Users] voicemail function

[Gary Carr]

I am not able to define the attach=yes/no setting on a per mailbox basis. 
The system uses the attach=yes/no statement under the [general] settings of 
the voicemail.conf file. I also tried removing the attach= statement from 
the [general] list.

Below is my mailbox config I have tried using
9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],attach=yes
Should'nt that work?

Thanks,
Gary
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[Asterisk-Users] rdnis

[Gary Carr]

Anyone have a example of how to setup RDNIS in *? 
To date we have been giving each voicemail user a individual DNIS but would like 
to consolidate all the numbers into one and just use RDNIS to route the 
call.



Thanks,


Gary

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Re: [Asterisk-Users] Tie web application to VOIP

[Gary Carr]

They have a free version coming out that raises the limit to 8 gig I 
believe.


Gary
Does Oracle have a decent-featured free version of their db software? That
was my original point, and where MS SQL 2005 is quite in the lead (limited
only to 1GB of RAM, 4GB DB, and 1 CPU).
-Michael
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[Asterisk-Users] Problems with DISA application

[Gary Carr]

I am testing DISA but can not dial out after getting local dialtone. The * 
server takes accepts my password, gives me dialtone and allows me to dial 
the digits then promply hangs up on me. CLI log shows

   -- Executing DISA(Zap/1-1, 1234|contextname) in new stack
   -- Accepting call from '9104025011' to '9105551212' on channel 0/1, span 
1
Dec 24 14:50:01 WARNING[1989798704]: app_disa.c:290 disa_exec: DISA on chan 
Zap/1-1 password is good
Dec 24 14:50:06 WARNING[1989798704]: cdr.c:286 ast_cdr_init: CDR already 
initialized on 'Zap/1-1'
   -- Executing Dial(Zap/1-1, Zap/g2/9104041000) in new stack
   -- Called g2/9104041000
   -- Channel 0/2, span 1 got hangup
   -- Hungup 'Zap/2-1'
 == No one is available to answer at this time
   -- Executing Hangup(Zap/1-1, ) in new stack
 == Spawn extension (contextname, 9104041000, 2) exited non-zero on 
'Zap/1-1'
   -- Hungup 'Zap/1-1'

My extensions.conf reads:
; Disa test
exten = 9105551212,1,DISA,1234|contextname
Any ideas?
Thanks,
Gary

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[Asterisk-Users] Linksys PAP2-NA not re-registering after registration expires

[Gary Carr]

We are testing some PAP2-NA ATAs from Linksys and I can use the device to 
make outbound calls as well as receive inbound calls when the ATA first 
connects to the * server. After a couple of minutes of being idle the ATA 
disconnects from the * server and will not take calls, but can continue to 
make calls. A sip show peers shows the ATA as being unreachable. Resetting 
the ATA or making changes to the registration expiry will force the ATA to 
reconnect but after another few minutes idle it disconnects again.

Anyone else seeing this with these devices or have a idea why this happens? 
We don't have this problem with Cisco7960s or X-Ten softphones.


Thanks,

Gary
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Re: [Asterisk-Users] VoIP Termination

[Gary Carr]

Why is IAX termination better?

Gary


So they offer termination via SIP for $0.013/minute?
Even better-- IAX termination :)
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Re: [Asterisk-Users] VoIP Termination

[Gary Carr]

So they offer termination via SIP for $0.013/minute?

Gary

Any since the per minute rates can be as low as $0.013/minute last time
I looked, you have to use a LOT of minutes before you spend as much as
you would have with that unlimited plan...
Regards,
-Dorn
p.s. I use both NuFone and VoipJet and am reasonably happy with both.

On Thu, Dec 16, 2004 at 01:53:44PM -0700, Mike Diehl (Encrypted email 
preferred) wrote:
On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
 On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) 
 wrote:
  One of the catches is that I often telecommute and sometimes I do 
  some
  side business; these practices violate many provider's acceptable use
  policies. So, I need a provider who doesn't care how I use the phone, 
  and
  one that works well with Asterisk.

 You've gotta be kidding, VOIP providers are trying to regulate who you 
 can
 call?  Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 
 over
 SIP, IMO it's just better.

Thanx, I will look into these providers.
This is an exerpt from Packet8's Terms of Use statement.  I've edited it 
for
space, but I've tried to retain the context:
--
PERSONAL USE. 8x8's Service Plans for residential subscribers that offer
unlimited minutes of PSTN calls (Unlimited PSTN Plans) are for the
reasonable personal residential use of End User only. End Users of 
Unlimited
PSTN Plans shall not use the Services for commercial or governmental 
purposes
or for profit or non-profit activities, including, but not limited to, 
home
office, business, sales, tele-commuting, autodialing, continuous or 
extensive
call forwarding, continuous connectivity, fax broadcast, fax blasting,
telemarketing or any other activity that would be inconsistent with 
personal
and residential usage. 8x8 reserves the right to immediately terminate or
modify the Services of any End User using Unlimited PSTN Plans if 8x8
determines, in its sole discretion, that End User is not using the 
Unlimited
PSTN Plans for End User's reasonable personal residential use.
--

Now I agree with their policy on fax-blasting, etc.  But according to 
them, I
can't use my own phone for charity work?  I work at a national lab; would 
my
wife be alowed to call me at work?  Or would the be a governmental 
purpose?

It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm
conducting a business with my phone, they can terminate my service, or
increase the price of it.
I'm trying to make an issue out of this because I think it needs to 
change and
I'm hoping people who are affiliated with these providers are reading 
this.
I was going to go with Packet8.  I was going through the final 
checklist
before subscribing when I came accross this fascist policy.

Sure, I can go with a business plan, but that would cost me $39.95. 
That's $5
more than I'm spending for an analog phone line!  Part of the reason for 
me
to go with VoIP is to become Quest Free.  But suddenly, Quest is 
starting
to resemble the Boy Scouts when compared to the types of usage policies 
I'm
seeing from some of the VoIP providers.

Sorry for the rant, but I hope you understand.
--
Mike
gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
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Re: [Asterisk-Users] Need an Asterisk Expert for a Project

[Gary Carr]

I'd like to have a system like that as well. I 
would be willing to chip in on the development.



Gary


  - Original Message - 
  From: 
  Paul Rodan 
  
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Cc: 'Commercial and Business-Oriented 
  Asterisk Discussion' 
  Sent: Friday, December 10, 2004 3:35 
  PM
  Subject: [Asterisk-Users] Need an 
  Asterisk Expert for a Project
  
  
  We have a customer that handles 
  the billing for a rather large company. Anyway, they have their phone system 
  through us, Cisco 79xx phones with Asterisk and such. They want us to build 
  them an IVR system that can interact with their billing system through XML and 
  read back information to the customer.
  
  They want their customers to be 
  able to call in, enter account numbers or credit card numbers or whatever, and 
  have the system read the balance or what’s been charged, etc. and give them 
  the option to cancel or increase their order size, etc. kind of like a bank 
  IVR system. Their billing interface is in XML, so I’m guessing we’d use 
  Asterisk’s AGI capabilities to call a script with certain parameters and that 
  script would post it to their system and get the results in xml and feed the 
  needed info back to Asterisk to be dynamically read off to the customer. I 
  personally don’t know how to do this, my developer and I could work on this 
  but the amount of time and energy it’d take for us to do it is just not 
  possible at this point in time. So they want me to find an expert for a 
  temporary project to make this system/setup work. I can provide more details 
  if you’re interested.
  
  Please email me if you or you’re 
  company can assist us in this project.
  
  Best 
  Regards,
  Paul
  
  

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Re: [Asterisk-Users] Wonderful Success with PAP2-NA

[Gary Carr]

where did you get them from?

Gary
- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 4:37 PM
Subject: [Asterisk-Users] Wonderful Success with PAP2-NA


Finally got authorized to purchase some PAP2-NA's from Linksys's.
Works like a charm with Asterisk. Web configuration has TONS of options 
and
looks nice.

Able to put line1 and line2 on seperate asterisk servers.
Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4
line ATA for $100.
-Matthew
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[Asterisk-Users] send to context when forwarded from a number

[Gary Carr]

Is there any way to send a caller to a certain context when the caller is 
forwarded from another pstn number. We are using * as a voicemail server for 
our cusotmers and we are currently providing each vm customer a did to send 
the caller to when their line is busy. I would like setup * to take the call 
and if possible tell where the call was forwarded from and send them to the 
vm context to leave a message.

Is that even possible?

Thanks,
Gary
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Re: [Asterisk-Users] Using Lucent/Ascend TNT as a PSTN Gateway?

[Gary Carr]

I would like to see those configs as 
well.



Gary


  - Original Message - 
  From: 
  Tim Connolly 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Wednesday, October 13, 2004 6:02 
  PM
  Subject: [Asterisk-Users] Using 
  Lucent/Ascend TNT as a PSTN Gateway?
  
  
   
  About a year ago, a couple * list users had working configs which used the 
  Lucent TNT as a gateway to the PSTN. Does anyone have working configs for both 
  ends they can post? My TNT accepts H323 call, and will call out when I dial 
  using it as a gateway, but I get no audio in either 
  direction.
  
   
  Any suggestions? No firewalls or NAT is involved in this 
  setup.
  
  

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Re: [Asterisk-Users] Seeking a VoIP Solution for a big company

[Gary Carr]

I don't understand your targeted market. Is your software available for 
people who have their own asterisk servers and if so why a limit on the # of 
usable ports?


Gary

Our already made solutuons are designed for just such scenarios.
Have a look at http://www.bicomsystems.com/products/C/SC/319/131/
Please contact me of the list for details.
Regards,
Senad J

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Re: [Asterisk-Users] Case studies for 120 simultaneous calls on IVR

[Gary Carr]

Hi-
I've run extensive load testing with both single and dual P4's and Xeon's
(all at least 2.8GHz), and I've got 6 installed IVR systems of this size in
various configurations.

Hmm, I was under the impression that it was impossible to run dual P4 CPUs. 
I thought Intel programmed instruction in the cpu to not post if 2 CPUs were 
found. What MB are you using to run the dual P4 system?


Gary

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Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9

[Gary Carr]

The RPMs had errors for me
After installing RPMS and running modprobe zaptel I get
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
register_chrdev_R07a6f6f0
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
remove_wait_queue_Rd7b46182
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
remove_proc_entry_R16f1fe81
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
__pollwait_Rb9575694
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
proc_mkdir_R68919af9
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
create_proc_entry_Rd11cc972
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol 
add_wait_queue_R1891d4b7
/lib/modules/2.4.20-31.9/misc/zaptel.o: insmod 
/lib/modules/2.4.20-31.9/misc/zaptel.o failed
/lib/modules/2.4.20-31.9/misc/zaptel.o: insmod zaptel failed

Going back to downloading directly.

Gary

On Thu, 2004-09-23 at 13:49, Chad Brown wrote:
Is anyone working on a Fedora Core 2 RPM?
Just download the RH9 src.rpm's and do a rpmbuild --rebuild on them.
--
Florin Andrei
http://florin.myip.org/
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Re: [Asterisk-Users] Linksys PAP2-NA

[Gary Carr]

Shite, I ordered some a few days ago from TD and they have my order on hold.

Gary

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It
installed pretty easily and has worked great so I went to order some more
of these units today.
When I logged into Tech Data this morning, the PAP2-NA was now marked as
discontinued and no longer available and only the PAP2 version was
available which is the Vonage branded version. :(
I saw someone on the list say that they heard from Cisco that these units
were not due out until Dec. Did Cisco/Linksys pull these units off the
shelves?
--
Eric Merkel
MetaLINK Technologies, Inc.
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Re: [Asterisk-Users] Linksys PAP2-NA

[Gary Carr]

This really chaps my hide.  The situation as it's been explained to me
is:  Apparently, too many *consumers* were accidentally buying the
PAP2-NA (unlocked) version and then complaining/returning them to
Linksys b/c they didn't understand that they need a service provider to
be able to place and receive phone calls.  Either the people intended to
buy the Vonage version, or they just didn't realize they needed a
service provider and ended up with a paperweight.  Linksys is pulling
all stock of these units back from the distributors and requiring that
they only be sold to ISPs who have the capability to provide VoIP
service.  If you've already got one, you're luckyI had my order
confimation but they hadn't shipped yet...My order was cancelled.
I hope to hear more from my vendor later today, I'll let you know what
else I can find out.
Well we're an ISP. Do you know the procedure for ordering them?

Gary
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Re: [Asterisk-Users] Linksys PAP2-NA

[Gary Carr]

Anyone confirmed a stocking vendor we can purchase these from?

Gary

Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally..  what 
about loading SPA-2000 or PAP2-NA firmware in the PAP2?  If it's the same 
hardware, there shouldn't be any reason not to try it.


Thats the first thing I'm going to try when we get our units.  I'll get 
them in a week or two and let you know.

--
Andres
Network Admin
http://www.telesip.net
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Re: [Asterisk-Users] Linksys PAP2-NA

[Gary Carr]

You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want 
to order some of these.


Gary

Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units 
and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel
Sent: Wednesday, September 22, 2004 10:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Linksys PAP2-NA

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It
installed pretty easily and has worked great so I went to order some more
of these units today.
When I logged into Tech Data this morning, the PAP2-NA was now marked as
discontinued and no longer available and only the PAP2 version was
available which is the Vonage branded version. :(
I saw someone on the list say that they heard from Cisco that these units
were not due out until Dec. Did Cisco/Linksys pull these units off the
shelves?
--
Eric Merkel
MetaLINK Technologies, Inc.
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Re: [Asterisk-Users] Linksys PAP2-NA

[Gary Carr]

I already called them. They don't have any and don't have a eta on a new 
shipment. Someone bought the 47 they had in stock this morning :-(


Gary

same here
On Wed, 22 Sep 2004 16:17:08 -0300, Bartosz Jozwiak [EMAIL PROTECTED] 
wrote:
I would love to have contact info for Bottom Line Tech also.
Then we do not have to go with all the trouble getting to them.

- Original Message -
From: Gary Carr [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 3:59 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA
 You have some contact info for Bottom Line Tech? We are a ISP/CLEC and
want
 to order some of these.



 Gary


  Eric,
  I was told by Bottom Line Tech that Linksys told them to pull all 
  units
  and
  stop all shipments unless there customer could prove they were and 
  ISP,
  which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
  John Millican
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Eric 
  Merkel
  Sent: Wednesday, September 22, 2004 10:07 AM
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Linksys PAP2-NA
 
 
 
  I receieved my first PAP2-NA yesterday from our distributor(Tech 
  Data).
It
  installed pretty easily and has worked great so I went to order some
more
  of these units today.
 
  When I logged into Tech Data this morning, the PAP2-NA was now marked 
  as
  discontinued and no longer available and only the PAP2 version was
  available which is the Vonage branded version. :(
 
  I saw someone on the list say that they heard from Cisco that these
units
  were not due out until Dec. Did Cisco/Linksys pull these units off 
  the
  shelves?
 
  --
  Eric Merkel
  MetaLINK Technologies, Inc.
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--
Michael Bielicki
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Re: [Asterisk-Users] Asterisk and Red Hat 9

[Gary Carr]

I am running a P4 2.8 with 1 gig of ram and 7200 
rpm IDE drives. Nobottlenecks as yet.




Gary


  - Original Message - 
  From: 
  Henry Devito 
  
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, September 20, 2004 8:25 
  PM
  Subject: RE: [Asterisk-Users] Asterisk 
  and Red Hat 9
  
  
  Thank you for all of 
  the replies. I would like to build a PBX with a 16 channel pri and 36 
  phones. What kind of processor and memory should I look at? 
  
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gary CarrSent: Monday, September 20, 2004 11:16 
  AMTo: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 
  and Red Hat 9
  
  
  I am running RH9 with a 4 port and 
  1 port ISDN cards. Not problems that I am aware of 
  yet.
  
  
  
  
  
  
  
  Gary
  
  
  

- Original Message - 


From: Henry Devito 


To: [EMAIL PROTECTED] 


Sent: Sunday, 
September 19, 2004 6:30 PM

Subject: 
[Asterisk-Users] Asterisk and Red Hat 9


Hi everyone, I’m a newbie 
to Asterisk. Will Asterisk run on RH9, easily or does it have to run 
on FreeBSD? Will the drivers for the Digium cards work on RH9? 




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Re: [Asterisk-Users] new ATA box for sale by Linksys

[Gary Carr]

They show as a active item on Tech Data's website but they don't have any in 
stock at the moment. They are available as drop ship from linksys.


Gary
- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 2:34 PM
Subject: Re: [Asterisk-Users] new ATA box for sale by Linksys


I had 2 senior level management people at linksys corp confirm that this
would not be possible until December. They both told me that they are
currently in development of a 'non-locked' version but that it would not 
be
in stores until December.

Did you find these PAP2-NA at Fry's as well? Online somewhere?
Thanks,
Matthew
- Original Message - 
From: Marty Mastera [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 1:27 PM
Subject: RE: [Asterisk-Users] new ATA box for sale by Linksys


Please explain how you got the PAP2 to work with another
carrier? I spent over an hour on the phone with 3 levels of
Linksys support staff and 2 levels of Vonage staff telling me
that the PAP2 CAN NOT be used on any other service except
vonage because they burn the vonage information into the firmware.
Please explain...
Matthew

Matthew:
When the PAP2 was first available, it was only sold as a Vonage locked
version (the same one that I and it sounds like you got...got nowhere
with it).  Since then Linksys has released the PAP2-NA which is not
locked to any particular service provider.  The part number is the
key...
Marty
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Re: [Asterisk-Users] Asterisk and Red Hat 9

[Gary Carr]

I am running RH9 with a 4 port and 1 port ISDN 
cards. Not problems that I am aware of yet.



Gary


  - Original Message - 
  From: 
  Henry Devito 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, September 19, 2004 6:30 
  PM
  Subject: [Asterisk-Users] Asterisk and 
  Red Hat 9
  
  
  Hi everyone, I’m a newbie to 
  Asterisk. Will Asterisk run on RH9, easily or does it have to run on 
  FreeBSD? Will the drivers for the Digium cards work on RH9? 
  
  
  

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Re: [Asterisk-Users] sip to sip calls thru asterisk

[Gary Carr]

That was coming from the register statements in the sip.conf file. Once I
removed those and restarted the sip clients everything started to work.




Thanks!



Gary

 It's not clear how you are making the call.

 You should be able to call directly from either phone to the other by
 dialing 5011 or 5012, respectively, if
 your context local  indeed contains the those extensions, which is not
 clear from your configuration excerpts.

 But it seems you are calling another user on carolina.net, who is
 registered with that provider from your asterisk,
 so the call will loop back to Asterisk.   SIP does not really have a
 good way to handle such loopbacks, and
 therefore you get the error.
 If you want to make this work you need to load a second SIP channel
 driver on your asterisk listening on a different port,
 The changes are not difficult.

 It also seems that both phones are sitting on the same IP address and
 port,  how can that be?
 Oh, I see the error message is actually coming from the sip phone, and
 it's because those phones
 have the same IP address, and therefore a loop is detected there.  Is
 this just ONE phone with two
 proxy-accounts or personalities?



 Gary Carr wrote:

  I have a test box setup and I can make outbound calls on the PSTN thru
  the diguim card, however I can not make a sip user to sip user call by
  dialing the extensions. I am getting the following error.
 
  -- Called cisco7960
  -- Got SIP response 482 Loop Detected back from 208.218.14.123
== No one is available to answer at this time
 
 
 
  CLI sip show peers
  Name/usernameHostDyn Nat ACL Mask Port
  Status
 
  cisco7960/5052   208.218.14.123   D   N  255.255.255.255  5060
  OK (1 ms)
  garycarr/5011208.218.14.123   D   N  255.255.255.255  5060
  OK (1 ms)
 
 
  sip.conf statements
 
  register = [EMAIL PROTECTED]/5011
  mailto:[EMAIL PROTECTED]/5011
  register = [EMAIL PROTECTED]/5052
  mailto:[EMAIL PROTECTED]/5052
 
  [cisco7960]
  type=friend
  host=dynamic
  nat=yes
  qualify=200
  dtmfmode=rfc2833
  canreinvite=no
  mailbox=5052
  callerid=Cisco 7960
  context=local
 
  [garycarr]
  type=friend
  host=dynamic
  nat=yes
  qualify=200
  dtmfmode=rfc2833
  canreinvite=no
  mailbox=5011
  callerid=Gary Carr
  context=local
 
  extensions.conf statements
 
  exten = 5011,1,dial(SIP/garycarr,20,tr)
  exten = 5052,1,dial(SIP/cisco7960,20,tr)
 
  Is this a possible nat issue? I can make a good call from behind the
  firewall doing sip to pstn so it seems 2 way traffic thru the firewall
  is working.
 
 
  I am still sifting thru the sip debug info but anyone has any ideas
  that would be great.
 
 
  Gary
 
 
 
 
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Re: [Asterisk-Users] Asterisk MIBS

[Gary Carr]

test message. No list messages received today.



Gary

- Original Message - 
From: Soren Rathje [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 5:08 PM
Subject: Re: [Asterisk-Users] Asterisk MIBS


Alagalah wrote:
 Hi,
 
 I was wondering if there are any Asterisk MIBS (specifically regarding
 call information) ?
 
 I noticed a post citing www.faino.org, but this site doesn't seem to
 exist anymore, and The Book v2 doesn't have any references to MIBS.
 
 Any pointers greatly appreciated.
 
 
 Keith Burns
 
 The dogs may bark but the caravan rolls on 

www.faino.it should be the correct link.

/Soren

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[Asterisk-Users] sip to sip calls thru asterisk

[Gary Carr]

I have a test box setup and I can make outbound 
calls on the PSTN thru the diguim card, however I can not make a sip user to sip 
user call by dialing the extensions. I am getting the following 
error.

-- Called cisco7960 -- Got 
SIP response 482 "Loop Detected" back from 208.218.14.123 == No one is 
available to answer at this time



CLI sip show 
peersName/username 
Host Dyn Nat 
ACL Mask 
Port Status

cisco7960/5052 
208.218.14.123 D N 
255.255.255.255 5060 OK (1 ms)
garycarr/5011 
208.218.14.123 D N 
255.255.255.255 5060 OK (1 ms)


sip.conf statements

register = [EMAIL PROTECTED]/5011register 
= [EMAIL PROTECTED]/5052

[cisco7960]type=friendhost=dynamicnat=yesqualify=200dtmfmode=rfc2833canreinvite=nomailbox=5052callerid="Cisco 
7960"context=local

[garycarr]type=friendhost=dynamicnat=yesqualify=200dtmfmode=rfc2833canreinvite=nomailbox=5011callerid="Gary 
Carr"context=local

extensions.conf statements

exten = 
5011,1,dial(SIP/garycarr,20,tr)exten = 
5052,1,dial(SIP/cisco7960,20,tr)

Is this a possible nat issue? I can make a good 
call from behind the firewall doing sip to pstn so it seems 2 way traffic thru 
the firewall is working.


I am still sifting thru the sip debug info but 
anyone has any ideas that would be great.


Gary

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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Gary Carr]

Did you check the dell outlet (refurbed with same warranty) or ebay? You can
typically purchase and ship same day from either of those.



Gary


 I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
PCI's,
 both brought to the rear panel - nice.

 BUT, I can't get the Dell's fast enough for this customer, so now I'm
 looking at the HP Proliant DL-320.

 Regards
 Scott Stingel


 Scott M. Stingel
 President,
 Emerging Voice Technology, Inc.
 Palo Alto California  London England
 www.evtmedia.com



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Re: [Asterisk-Users] Cisco 79xx series IP phones

[Gary Carr]

Can anyone send me the sip images for the 7960g? I have 2 I want to test but
need them to be sip.



Thanks,


Gary


I've tried a *lot* of phones with Asterisk, and thus far, the Cisco's
are by far the best I've used.

Brian D'Arcy


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shawn Parker
Sent: Friday, August 13, 2004 9:31 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 79xx series IP phones

I got a call from our Cisco rep today saying that they couldn't sell
just phones to anyone because if my ethernet isn't to exact spec...
then they won't work at all.  I've read over the Wiki documentation and
it seems that the 79xx series phones work with Asterisk.  They told me
that without a Cisco phone system in place or a Cisco router or switch,
then the ethernet wouldn't work with the phones.  Is this true, or is it

someone just trying to sell me a Cisco system?

I don't see how my use of a Planet or Netgear switch would alter the
spec of my ethernet to cause a IP phone to fail.  Seems far fetched to
me.  I've never had any other problems mixing Cisco equipment with other

product lines.

Does anyone have any knowledge or experience to give me dealing with
Cisco 7902G and 7905G IP phones and getting them to work on a lan with
Asterisk when *not* using other Cisco hardware?

Cheers,

Shawn
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[Asterisk-Users] uniden phones

[Gary Carr]

Who are the US wholesalers selling the uniden phones?


Thanks,


Gary


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Re: [Asterisk-Users] Using Cisco SIP Phones with Asterisk

[Gary Carr]

Per that url you have to get a support contract to change the phone from
skinny to sip and you got the runaround trying to contact Cisco to get
access to the sip images.



Gary




 Which hurdles are you talking about specifically? These phones work
 great with asterisk (as long as you install the SIP image on them).

 mitchel

 On Wed, 4 Aug 2004 15:57:11 -0400, Gary Carr [EMAIL PROTECTED] wrote:
  Are they still hurdles using Cisco phones with asterisk as mentioned at
  http://www.voip-info.org/wiki-Cisco+Phones ?
 
  We are looking for some cisco phones to test with.
 
  Gary
 
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[Asterisk-Users] voicemail attachment setup per user

[Gary Carr]

Is it possible to set the attach= setting on a per user or per context
basis? We want to give our users the choice of no email notfiication, email
notification with no attachment, or notification with attachment.



Thanks,


Gary


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Re: [Asterisk-Users] Rodopi Billing

[Gary Carr]

 That sigh will turn to cursing after a couple of months. We currently use
 Rodopi, have for 10 years but the inflexability is too much to deal with
 anymore so we are moving away from it.
 
 To what?  I am also a cursed Rodopi owner. :-(
 
 Tom


We bought the source code to wirebill and are building our own platform.



Gary


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[Asterisk-Users] Using Cisco SIP Phones with Asterisk

[Gary Carr]

Are they still hurdles using Cisco phones with asterisk as mentioned at
http://www.voip-info.org/wiki-Cisco+Phones ?



We are looking for some cisco phones to test with.



Gary


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Re: [Asterisk-Users] Rodopi Billing

[Gary Carr]

Mostly CLEC stuff like CDR imports for specific ILECS  LD carriers as well
as some ISP stuff like redirecting past due accounts to a payment page as
well as any other stuff we may need. We plan to offer it to other service
providers as a ASP model and for purchase.



Regards,


Gary


 WireBill looks interesting. You mentioned that you are using the source
 code to build your own platform, but how does it hold up on its own? Can
 I ask what it can't do that requires you to build your own?

 Thanks,

 - Darren

 On Wed, 2004-08-04 at 08:14, Gary Carr wrote:
   That sigh will turn to cursing after a couple of months. We currently
use
   Rodopi, have for 10 years but the inflexability is too much to deal
with
   anymore so we are moving away from it.
  
   To what?  I am also a cursed Rodopi owner. :-(
  
   Tom
 
 
  We bought the source code to wirebill and are building our own platform.
 
 
 
  Gary
 
 
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Re: [Asterisk-Users] Rodopi Billing

[Gary Carr]

While we have not integrated the asterisk CDRs yet it should not be a
problem to do. We our building a billing system for ISP/CLECs that will do
what you want. If you want more information you can contact via email to
[EMAIL PROTECTED] or by calling 910.402.5010



Regards,


Gary Carr
President/CEO
705A Wesley Pines Rd.
GSC Telecommunications, Inc.
Lumberton, NC 28358
Phone: 910-402-5011
Fax: 910-618-9027
Check us out at: www.gsctele.com



 Well, can anyone recommend a full featured ISP billing system that would
 handle VOIP/Asterisk?

 - Darren

 On Tue, 2004-08-03 at 11:09, Brian D'Arcy wrote:
  On Fri, 30 Jul 2004, Darren Bentley wrote:
 
   Hello,
  
   Has anyone used Asterisk in conjunction with a billing system like
   Rodopi? Is the Rodopi VOIP module worth getting, or can radius be
  used?
 
   I suffered with Rodopi for three years in a previous life. Avoid it
  like
   the plague.
 
  OMG.. I had to support a rodopi installation myself for 2 years..
  Closest I've ever come to suicide.  While I have not managed another
  system but RODOPI, I have to say, there must be better.
 
 
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Re: [Asterisk-Users] Rodopi Billing

[Gary Carr]

That sigh will turn to cursing after a couple of months. We currently use
Rodopi, have for 10 years but the inflexability is too much to deal with
anymore so we are moving away from it.



Gary

- Original Message - 
From: Ejay Hire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 7:54 PM
Subject: RE: [Asterisk-Users] Rodopi Billing


 Thanks for the vote of confidence guys.  We just bought an
 ISP that uses rodopi exclusively for Accounting and Billing.


 ...sigh...

 -e

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf
 Of
  [EMAIL PROTECTED]
  Sent: Tuesday, August 03, 2004 12:15 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Rodopi Billing
 
  On Fri, 30 Jul 2004, Darren Bentley wrote:
 
   Hello,
  
   Has anyone used Asterisk in conjunction with a billing
 system like
   Rodopi? Is the Rodopi VOIP module worth getting, or can
  radius be used?
 
  I suffered with Rodopi for three years in a previous life.

  Avoid it like
  the plague.
 
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Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch

[Gary Carr]

Hi, which IP Centrex setup are you using?



Gary



 I am using asterisk as a voicemail server for our IP Centrex SoftPBX.

 Umar.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten
 Sent: 09 July 2004 22:46
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5
 softswitch


 when you say you have integration what exactly do you mean?  are you using
 asterisk as the voicemail system for a class 5 switch?

 On Friday 09 July 2004 15:45, usedcanon wrote:
  I have integration. Asterisk is upto the task however you may need to do
  some work arounds.
 
  Umar.
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten
  Sent: 09 July 2004 20:51
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] using asterisk voicemail with a class 5
  softswitch
 
 
  anyone have any idea on the compatibility of asterisk voicemail with a
  class 5
  switch that can do SIP (in particular the MetaSwitch VP3500)?
  --
  Chad Whitten
  Network/Systems Administrator
  [EMAIL PROTECTED]
  601-944-4801 Phone
 
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 -- 
 Chad Whitten
 Network/Systems Administrator
 [EMAIL PROTECTED]
 601-944-4801 Phone

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Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch

[Gary Carr]

I am not sure of your answer but we are looking to integrate * with our
class 5 switch to provide Voice Mail services to our subscribers. If anyone
here has a interest in performing this integration from a
consultant/contract basis please email me offline.



Thanks,


Gary Carr
President/CEO
705A Wesley Pines Rd.
GSC Telecommunications, Inc.
Lumberton, NC 28358
Phone: 910-402-5011
Fax: 910-618-9027
Check us out at: www.gsctele.com



 anyone have any idea on the compatibility of asterisk voicemail with a
class 5
 switch that can do SIP (in particular the MetaSwitch VP3500)?
 -- 
 Chad Whitten
 Network/Systems Administrator
 [EMAIL PROTECTED]
 601-944-4801 Phone

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Re: [Asterisk-Users] TDMoE Question

[Gary Carr]

Rad's TDMoIP uses DSP chips on each end of the link to compress the data.



Gary



 Just a Question. I would like to know if TDMoE follows specifiaciones of
 TDMoIP RAD protocol that says that there is a compression of 16/1 when
 you do TDMoIP.



 Manuel Marin Garcia
 TRANSTELCO S.A. DE C.V.
 Campos Eliseos 9050 B4 â? Cd. Juárez, Chih. 32452 - México
 Oficina: +52 656 692 11 09 â? Fax: +52 656 692 1112 - Celular: 915 727
 6141
 http://www.transtelco.com.mx

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