Displaying 11 results from an estimated 11 matches for "voiceglo".
2003 Dec 01
8
VoiceGlo
Hi,
VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll
Take a loock on http://www.voiceglo.com/
The softphone is IAX :)
Best regards,
Chris HARIGA
Techselesta Inc.
http://www.techselesta.com/
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2005 Jan 05
2
Glophone/Voiceglo and Asterisk
<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P>
<P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P>
<P>I've tried copying the config in this listing with...
2003 Dec 23
0
Voiceglo SIP configuration
...g just seems to prevent the call from
going through.
Username is your 10-digit phone number.
Password is in the .reg file they sent you via email. I signed up for the
USB phone, so I don't if they send a .reg file if you went for the MTA.
sip.conf
register => 1234567890:password@myphone.voiceglo.com
[myphone.voiceglo.com]
type=peer
username=1234567890
secret=password
nat=no
host=myphone.voiceglo.com
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=200
restrictid=no
fromuser=1234567890
fromdomain=1234567890.voiceglo.com
extensions.conf
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTE...
2004 Jan 30
1
Cameron Palmer / voiceglo
I found a message in the archives from Cameron Palmer on 23 Dec regarding
his voiceglo SIP configuration. Unfortunately (for me), the archive has
his email address removed.
So, Cameron -- or anybody else using voiceglo with their * box -- please
reply to me so that I can get your email address and ask you a question
about your setup.
Thanks,
Greg
2004 Jan 22
0
voiceglo.com and dtmf
Hello all,
I've been trying to get a simple PBX up and running with
asterisk. I decided to sign up with Voiceglo so I could have a PSTN
gateway. The problem is that I can't seem to get Asterisk to handle
dtmf decoding reliably. I tried inband and the rfc decoding. inband
tried to work and the rfc mode didn't do anything. By try to work I
mean that it rarely properly decoded the number correctly....
2005 Oct 04
12
Sprint Nextel sueing over VoIP patents
Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing
on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP
technologies. Does anyone know which ones this article is talking
about, and if so does asterisk have any of those features?
The reason I am asking is that the article is vague, Vonage uses...
2004 Jun 01
2
BroadVoice usage?
...set up so I can get access to their servers.
However, something's still not quite right when I use the parameters.
It looks like our Asterisk server is registered with their server and I
believe I have all the other entries set up correctly (we use other services,
too, such as iConnectHere and Voiceglo so I'm pretty sure I've got my
BroadVoice set up correctly.) "sip show registry" shows my * server
as registered with the BroadVoice IP address.
When I try to make an outbound call, there is no answer from the
BroadVoice server (proxy.boradvoice.com) to Asterisk's INVITE.
Wh...
2004 Apr 27
12
VOIP providers
Is anyone signed up with Vonage and using an asterisks box??
Also what VOIP providers would anyone recommend?
--
James Moran
Potential Technologies
http://www.potentialtech.com
2005 Mar 04
2
Asterisk box and verizon calling it
I set up an asterisk box with a broadvoice sip connection for incoming
connections
it works great when I use a cell phone, vonage line, calling card to
call the asterisk box, but when I try to call it from our verizon land
line it is busy and asterisk logs do not show incoming call.
Any ideas on what the issue is?
Thanks!
Randy
2003 Dec 23
2
Fw: perl database get
i mean AGI->database_get()
----- Original Message -----
From: "Muhammad Nasim" <muhammad@telappliant.com>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, December 23, 2003 6:41 PM
Subject: Re: [Asterisk-Users] perl database get
> I've used both the syntax you have given and the perl module.
AGI->getvar()
> returns nothing for arguments that work from
2004 Jan 26
0
I need some clarification on DTMF
Hello all,
I have a SIP provider that is doing the PSTN bridging for me
(voiceglo.com). I understand the inband, info, etc for DTMF on *my*
network, but what about what they're sending me? Is there anything that
could prevent them from sending me DTMF information? The Asterisk
system is currently NAT'd (no special rules, just a Linux router) and
incoming and outgoing...