launch
new services until we can get Asterisk installed and running in our
location.
Please let me know if you can provide this service or if you know of a
company that can do it for us.
thanks!
Sam Michelson
-Original Message-
From: Lubomir Christov <[EMAIL PROTECTED]>
To:
unsubscribe
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ports should i specifiy?
>
> Simon
>
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If you can, try giving the highest
enerated. I
cannot confirm yet if it has fixed my dtmf talk off problems, but I have
not had any problems navigating through company ivr's (of course I
didn't before either.)
Sam
Christopher Gray wrote:
> Hello:
>
> I need to be able to reliably send out touchtone to any ca
Christopher, did you receive the email that I sent to your yesterday?
It was delivered Jan 23 20:47:31 -0600. Maybe it went to your junk box..
I will try again.
Sam
Christopher Gray wrote:
> Hello:
>
> Yes, DTMF can be a problem on the phones themselves as Sam observed, and
> i
Muiz Motani wrote:
> I am experiencing choppy sound when I bridge from a SIP peer to an IAX
> peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
> experiencing choppy sound from the SIP peer to the IAX peer but not
> vice-versa. I know that this is not a bandwidth issue because
ay.
http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html
Sam
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f time
to get everyones schedule to line up, I don't want to go through the
trouble if I will just be disappointed.
Thanks,
Sam
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Actually I think they will all be calling in using regular pstn phones
and cell phones.
Sam
Al Baker wrote:
> The 2 big questions are:
> -Are all participants using QoS end to end ?
>
> -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
> the work on th
coming from asterisk and not the ata. And all the dtmf modes
are rfc. Any one have any tips to further trouble shoot this?
Sam
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Somewhat off topic but does anyone know if the price for the license
will go down in the future? It seems strange that I can use skype for
free on my computer but to put it on asterisk cost $66...
Sam
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Any suggestions on how to go about
this?
so person calls, recording: "press 2 to call
cell phone", user presses 2, call forwards to my cell phone.
Thank you
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To
Hi,
If I setup an IP PAX gateway to handle VoIP calls to a traditional phone
line, I am wondering how each VoIP call to the PSTN connection get
charged by a local Telecom.
Thanks
Sam
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Jean-Michel Hiver wrote:
Sam a écrit :
Hi,
If I setup an IP PAX gateway to handle VoIP calls to a traditional
phone line, I am wondering how each VoIP call to the PSTN connection
get charged by a local Telecom.
I am not really sure to understand the question. But assuming you are
ill probably have to discuss charging with your Telco.
PaulH
_
Thanks for the answers. I really appreciate that. It may be better for
me to talk to local Telco for further price negotiation.
Thanks
Sam
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harge from IP PAX
(or PSTN) gateway to end phone. Assuming that there are 1000 VoIP
calls thru Telco's PSTN to end phones, how will these calls get
calculated? is the charge will be per-call basis?
Sam, I am still unsure to understand your question :-/
How much your telco is going to cha
ine. I thought I can send calls out thru a normal home
telphone line. If this is the case, I will just need to pay each VoIP call
to phone line at 20 - 30 cents / call.
Thanks
Sam
> PaulH
> ___
>
___
--
anks for the answers. I really appreciate that. It may be better for
>> me to talk to local Telco for further price negotiation.
>>
> Going through a VOIP termination service is also good for testing.
>
When going thru a VoIP
Hi,
Is there any specific made embedded hardware designed VoIP software or
Asterisk? I want to build a router that have VoIP enabled, so that I can
use it connect to a VoIP ISP.
Thanks
Sam
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Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
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Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
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mustardman29 wrote:
Not a lot to go on sam. What do you want to do? If you just want to play
or have very minimal requirements then get a soekris NET4801 board, CF and
install Astlinux.
http://www.soekris.com/net4801.htm
-Original Message-
From: sam [mailto:[EMAIL PROTECTED
Jim Houser wrote:
http://gumstix.com/waysmalls.html
Thanks for your link. how to build asterisk into this hardware?
Thanks
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sam
Sent: Friday, March 31, 2006 8:01 AM
To: asterisk-users
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps something needs to be looked into...
Regards,
Sam
#x27; dialplan option. Simply
changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to
'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.
But this seems really hackish. Is this the right/only way? Or is just
having a provider and mismatch
On 08/21/2015 12:52 AM, Sam wrote:
Hello,
I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains b
detecting
the hangup and ending the connection?
Sam
smime.p7s
Description: S/MIME cryptographic signature
Zaptel was the version from about 4 days ago when I sent this message, I
updated again yesterday night
Sam
Quoting Martin Pycko <[EMAIL PROTECTED]>:
> How old is your zaptel code ?
> Mark recently increased some timer for that.
>
> Martin
>
> On Wed, 18 Jun 2003, Sam
here --
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, June 19, 2003 11:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoiceMail recording dialtone
Well experiment yourself with the code.
in wc
OK, I tried upping it to 2000.. See if that changes anything
I still don't understand why it would end up directly in voicemail if it
picked the line back up instead of calling extension "s" again if the
telco's hangup signal was interpreted as a ring?
Sam
-Origin
Title: Message
Hi,
I am new to Asterisk
and was wondering if Asterisk has the ability to act as a protocol
converter.
I have an H.323
network and I want to know if Asterisk can convert the signaling to SIP so I can
send it to SIP Addresses?
Thanks for your
help!
regards,
Sam
is situation. Any tips or pointers to online docs
would be appreciated.
Thanks,
Sam
P.S. Thanks to Jsmith for the fast, simple answer to my last question
re: version number in CVS not updating.
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sterisk running it will die. This simplifies life for asterisk.
Mark, if you think this patch is stable feel free to apply it... You have
my waiver already.
Sam
moh_sox.diff
Description: Binary data
smime.p7s
Description: S/MIME cryptographic signature
I have a working MP3 decoder in a thread, using libresample and
libmp3lame, but I'm not really happy with it yet Not sure about the
legalities but if anybody wants to try this work in progress just let me
know
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
Hi all,
I'm having a problem with * being very finicky about the length of
DTMF key-presses during menus, voicemail, etc. Basically, short (<100
ms) tones are ignored, anything between 100ms (or so) and about 300ms
is correctly detected, and anything >300ms is interpreted as multiple
presses o
be handled...
Actually, we should be able to get a pretty good idea on that by using two
gigabit interfaces and VoIP?
Sam
Quoting Andy Hester <[EMAIL PROTECTED]>:
> > >
> > >I talked to Imagestream this morning about the possibilites. Their lead
> > >engineer said that t
avn't ever been reversed so I can neither confirm nor deny this.
Sam
Quoting Dan <[EMAIL PROTECTED]>:
> Hi,
>
> I have a FAX machine connected to a TDM400 card FXS port.
> When I receive a fax call through X100 and transfer it to that extension,
> the FAX machine dis
Also, if you ONLY run * on the system, you can lock it down so that the
security bugs are pretty much non-exploitable... Ipchains/etc. You don't
even HAVE to run ssh or any remote management if you want to to be just
like a regular PBX system
Sam
-Original Message-
From: [
If you buy the codec, it will do conversion...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of NetOne
Administrator
Sent: Friday, January 16, 2004 3:24 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] G.729 quiestion
Hi all!
If i purchase the G.729 co
You need to install the "mysql-devel" rpm if you use redhat
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Sunday, March 28, 2004 2:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Error installing/compiling cdr_m
* listens for fax tones as soon as you "Answer()" the line. If you Answer
the line before ringing the local lines, it will actually detect fax tones
while in the Dial statement.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
order, so it
is not an Asterisk problem.
Please Help!
Thanks,
Sam Bacsa
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No, the voice mail runs without any
hardware.
Check out http://www.voip-info.org/ for information
about implementing voicemail into Asterisk.
- Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
PTCHENSent: Sunday, April 04, 2004 8:18 PMTo:
[EMAIL PROTECTED]Subject
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Tuesday, January 15, 2008 3:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Sunday, January 20, 2008 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk
It is more economical to get a hardware GSM Gateway from places like
cyber-telecom.net and then plug it in a X100P
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Monday, January 14, 2008 8:43 PM
To: asterisk-users
may be I should keep my month shut..
But hey I though a mailing list is trying to get other users helping each
other.
There is no way I cannot see my post being non constructive ..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent
se in. Then they are setup just like a normal phone line
Try to look at the voip wiki for x100p config
2. get a 8 ports fxo voip gateway and 8 ports gsm gateway then put those
together. They end up working the same way.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
Sam
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent: Wednesday, January 30, 2008 10:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk gateway
Hello everybody
Well I think you need a GSM Gateway
You can find some info on cyber-telecom.net
For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a
X100P or something similar then it would be very economical.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Well alternatively you can look up straight forward gsm voip gateway.
Like CT-375
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, February 07, 2008 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am using about 70 Grandstream GXP2000 phones on 1.1.5.15 code with *
1.4.16.2 and have not experienced any of these issues. The one thing
that I would suggest is make sure that you are using RFC2833 for you
DTMF Mode. I was originally using INFO and ran into some strange issues
with dropped cal
Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have
not seen the issue that you are describing, but I had some very strange issues
like call hang-ups when I was using INFO. After switching the issues were gone
and I have had no further troubles.
Hope this helps you.
I know this is a bit off the thread
But I am trying to see if anyone in here know how to config a AS5300 with 2
T1.
Please contact me off list if you can give me a bit of help
Sam Tam
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el GSM gateway, and have the
ability to make more than 1 call at once through it.
Thanks
-Kev
Sam Tam wrote:
> Try cyber-telecom.net
> May be get a X100P with a CT-G1000 or G2000
>
> Sam
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTE
Hello Sebastien
First this is not the way to unsubscribe
Second you can do better with your spelling too.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastien
Mortier
Sent: Sunday, March 16, 2008 4:30 PM
To: Asterisk Users Mailing List - Non
It seems to be hard to find around the web. If anybody got a copy please
feel free to drop me a copy via email.
Many thx
Sam
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I have been trying to source a trixbox ce manual in pdf but if anybody can
point me to the right direction then it would be good.
Sam
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I am in the process of implementing a new ISDN pri and have a couple of
questions. This is a full 24 channels (23 B and 1 D) delivered over a T1
interface. The interface looks good and is not showing any errors. Any help
that you can provide would be greatly appreciated.
1) What switchtype
kup(${EXTEN:2})
exten => _**XXX,n,Hangup()
[BLF]
; Defines a BLF Hint for phones
exten => 212,hint,SIP/sam
-snip-
SIP.CONF
-snip-
[sam]
type=friend
username=sam
fromuser=sam
callerid=sam
host=dynamic
dtmfmode=RFC2833
disallow=all
allow=ulaw
call-limit=20
subscribecontext=BLF
Thanks in ad
risk-users] Pickup Command not working
On Tue, Nov 06, 2007 at 05:04:50PM -0500, Lutgring, Sam wrote:
> When I execute a pickup on a ringing phone I get CALL FAILED REASON
> CODE 603. I am dialing **212 with the following config. Anyone have
> a suggestion?
I am not sure, but in the cont
n't know why it is not doing the
pickup.
Thanks for the help though.
-Original Message-
From: Baji Panchumarti [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 07, 2007 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Lutgring, Sam
Subject: Re: [asterisk-u
Why not simply store voicemail local so there are no issues if the VPN
goes down. Then set up your dial plan at each site to allow the PSTN
access to your remote (other site) extensions. You can use IAX to trunk
a "PSTN" call just like you can a local caller, just give them access to
the context.
phone.
Here is what it looks like:
SIP.CONF
[sam-X-1433]; This is my X-lite phone
type=friend
username=sam-X-1433
-SNIP-
[sam-G-1433]; This is my desk phone
type=friend
username=sam-G-1433
-SNIP-
EXTENSIONS.CONF
exten => 1433,1,Dial(SIP/sam-G-1433&SIP/sam-X-1433,22,Tt)
Hope you f
Is there a way to allow a user to dial an extension after listening to
your voicemail instead of leaving a message? Example would be the big
boss is on vacation and changes his out message to say "you can reach my
assistant at by dialing 1234 now or leave me a message".
Thanks in advance.
__
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI) the
Caller ID Name is displayed correctly, but the Caller ID Number seems to
be empty. My Gr
Has anyone experienced the situation where you receive a
PRI_EVENT_PROGRESS message from a PRI that is then sent to a SIP channel
where the SIP client (tried 2 different phones/manufactures) never
acknowledges, Asterisk resends the message two more time and then begins
hanging the call up?
This
I have mine set up to ring a group of designated phones. Each one of
those phones has a dedicated line button that subscribes to their
particular account in the group. This way when the phone rings the user
KNOWS that it is the main building number that is ringing.
-Original Message-
Fro
I am having an issue with the CallerID Number not being passed to my
phone in the SIP packet. The CallerID Name is passed just fine and
displayed on the phone with no issue. I have done a NoOp() in my
extension.conf and successfully seen both the CallerID name and number
correctly. So that leads
this code at checkout and enjoy 15% off on the total order
excluding delivery charges.
Kind Regards
Sam Tam
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I think what you need is a GSM Gateway
You can find a cheap one at cyber-telecom.net
The model you should be looking for is CT-G1000 or 2000
Plug the SIM in there and it will give you a RJ11 telephone port where you
plug into something like X100P then you are ok to go..
Sam
-Original
Has anyone had any experience using a modem through the Asterisk system?
I have some technical support personnel that need to use a computer
modem to connect to a remote system for troubleshooting. Is there a SIP
compliant gateway that will support a modem connection at decent speeds
(minimum of 2
listen to them too/
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Anness
Sent: Tuesday, October 21, 2008 3:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How Secure Is Asterisk
I am sure this has been
Secure Is Asterisk
lol
On Mon, Oct 20, 2008 at 3:34 PM, Sam Tam <[EMAIL PROTECTED]> wrote:
VPN IP phone?
Then firewall up the asterisk to disable any outside access and
place the
vpn server with the asterisk in a locked cabinet .
Sure that wil
annoying.
So wondering is there anything I can do to make it work with Asterisk or am
I good to send back to exchange another item?
Sam
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Yes I know too.
Is there anyway to make it work with asterisk without using Callmanager?
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons
(US)
Sent: Friday, January 23, 2009 5:10 AM
To
And I assume no one know when they will have a SIP firmware for it too
right?
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, January 23, 2009 5:44 AM
To: Asterisk Users Mailing
Well does it matter if the asterisk server is not located in the same
network?
I am willing to spend a bit of cash to get someone help me to set it up .
Since I need it quite done before end of this month
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi
I am no expert in the cisco phone
Do you have time to help
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico
Santulli
Sent: Saturday, January 24, 2009 12:59 AM
To: Asterisk Users Mailing List
As far as I know all POE switches are quite noisy, they need to cool the
extra power consumed by the POE and hence they will run warmer than other
switch.
I know Cisco, 3COM, are very noisy but you can try other cheaper brand like
levelone or other to see if they have fans inside the switch.
Good
Hello I need someone to install Chan_SCCP for me and get it working on
Elastix with Cisco 7937
Interested party please msn me on sam__tam AT hotmail DOT com or email me
back
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0 per port with a max of 16 ports in 1
chassis.
Sam
-Original Message-
From: Matteo Brancaleoni [mailto:[EMAIL PROTECTED]
Sent: Monday, October 30, 2006 12:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP GSM Gateways
Hi,
On Sun, 2006-
Happy New Year …..
Sam
_
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Monday, January 01, 2007 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Happy 2007!!!
Its the new year. Cant we all be semi nice for atleast a lil bit
Drop me an email
I know some GSM Gateway that has a direct serial port for SMS>
Sam
-Original Message-
From: Jon Pounder [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 14, 2007 10:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMS from Aster
via normal post office
mail un-insurance. If you prefer insurance and express service, I can send
it via DHL for £40.
More info can be found on cyber-telcom.net
Sam
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asterisk-users
Why if you have 50 operator then I would even consider using dual server
running backup
So the idea of using vmware may really be very risky, let alone not talk
about performance issue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad
Jordanovic
Sent:
There will be a slight of delay on writing files but not really a performace
issue at all.
You will hardly notice.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ronald ramos
Sent: Saturday, May 24, 2008 5:30 PM
To: Asterisk Users Mailing List - Non
Try adding you context in that the phone is subscribed to. I had some issue
with this because if you do not specify the context it defaults to “default”
and has trouble finding the phone correctly. If you watch your debug very
closely I you should see it try to pick the phone up in the wrong c
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are
Well to be honest, our experience with asterisk never works with under NAT.
if you got DMZ then it will otherwise don't hold your breath for it.
If you want to use it as a production server
Your option is 1. Get a Real IP
2. there is no 2 really just get an ReaL Public I
I use the GXP 2000 and have had no issues with them keeping the correct
time. I run my own NTP server and point the phones to that source. As
I stated, this is working very well for us. A couple of simple things
that I would suggest you check:
1) On the Basic tab make sure that you
I am after someone to help me to config H323 on asterisk if possible since I
am far too busy stuck on another project. Interested parties please msn me
on sam _ _ tam AT hotmail.com please take out all space and change AT to @
If you are unsure then you can always email me with your contact via
We have got that for $10 USD setup and $25 USD per month
If you are interested please email me back
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Friday, July 18, 2008 9:48 PM
To: Asterisk Users Mailing List - Non-Commercial
No problem, you know you can always email us again if you have any other
requirement in VoIP>
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Saturday, July 19, 2008 12:05 AM
To: Asterisk Users Mailing List - Non-Commerc
Dear all
Does anyone know where I can find some good quality Russian language voice
file for calling card?
Thanks in advanced?
Sam
___
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AstriCon 2008 - September 22 - 25 Phoenix
If I am right I think you will find that you will not have enough power to
run 4e1 with g729 codec on little 1950..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Friday, September 26, 2008 3:08 AM
To: Asterisk Users Mailing List
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do test
at every interval too
If you know something like this please enlighten me.
Sam
You actually using that steve?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Saturday, September 27, 2008 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator
On
for inbound or outbound.
I can get you free inbound test DID. LMK
Jai
www.didforesale.com
On 9/27/08, Sam Tam <[EMAIL PROTECTED]> wrote:
> Hello everyone
>
>
>
> I am trying to look for a free test call generator that will get me some
> stats like PDD, ASR and call quali
Why not swap it all with just IP phone?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vieri
Sent: Monday, September 29, 2008 4:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ATA for large networks
Hi,
I would like to know if
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