Why would anyone pay $19.95/mth for termination only when you can pay the
same price and get unlimited INCOMING (DID) and outgoing with numerous SIP
providers?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 10
arket research,
we want to know what people want to buy and what they want to pay for it.
Also, IAX2 is, imho, a better protocol than SIP over udp especially for
users with standard broadband connections to significantly oversubscribed
ISPs, but I'm sure that will spark some discussion...
-Joe
Isn't Global Crossing Based in Bermuda?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Thursday, May 12, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT:
X-Ten is beta tesing a new softphone for Linux and could use some beta
testers. If interested, please contact: Neil McGuigan, [EMAIL PROTECTED]
I am currently testing it and it appears every bit as good as the
Windows version.
___
Asterisk-Users mail
It appears that it just supports SIP.
[EMAIL PROTECTED] wrote:
X-Ten is beta tesing a new softphone for Linux and could use some beta
testers. If interested, please contact: Neil McGuigan, [EMAIL PROTECTED]
I am currently testing it and it appears every bit as good as the
Windows version.
Will /
Why can't Broadvoice just LEAVE WELL ENOUGH ALONE!! Now, after
applying these new variables, I can't receive INCOMING calls. Sheesh,
what a bunch of BS!! Now we have to spend another weekend fixing what
BV screws up.
Dan Weber wrote:
Today, We have added INVITE Authentication. This seem
changes and start working again when I
remove your changes?
Also, insulting your customers is not the way to keep them. Or maybe BV
wants to get rid of Asterisk users.
Dan Weber wrote:
They are completely unrelated. Maybe you should read instructions.
Dan
On Sat, 5 Mar 2005, Mike Matthews wrote
Thanks for sending this. However, I literally cut and pasted your
examples (with my sip credentials) and incoming calls still go
automatically to BV Voicemail. Using sip debug shows that the call never
hits my * box. Thank anyway...it was certainly worth a try.
Marios Andreou wrote:
Its worki
Thanks. Adding those lines appears to have fixed the problem. I'll
just hold on til the NEXT TIME Broadvoice decides to make a change.
Thanks again.
Bartosz Wegrzyn - asterisk wrote:
I don't know what is wrong with the Broadvoice, but for me everything
works fine. I used the setup they provi
I have to say that I have had excellent service and support from
Livevoip as well. Every support issue I have had has been handled
inside of 2 hours and that includes weekends. I have had a Sr. VP there
call me at midnight on a Saturday night to work out a problem. They
have DID's in MANY ar
This works for me both incoming and outgoing w/Sipphone. Note there is
NO username, secret entries in the peer definition. I am using * vers 1.05
register=1747nnn:[EMAIL PROTECTED]/1747xxx ;
note:extension in extensions.conf matches for incoming
[sipphone]
type=peer
host=proxy01.sippho
Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
aut
You will probably get it Monday morning. Thats what happened to me when I
ordered on a Friday noght.
- Original Message -
From: "David Uzzell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, March 13, 2005 8:10 PM
Subject: [Asterisk-User
I purchased the g729 codec from Digium. Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall? Perhaps by ba
For the record here, I am quoting from an email from Digium on the subject:
"You will need to backup /var/lib/asterisk/licenses. You will also needto backup the codec_g729 and format_g729 in your/usr/lib/asterisk/modules/ directory. The ethernet cards in yourmachine cannot be changed. Otherwi
Have each phone peer reference a different context. Have the different
contexts set up to use the particular service for that user.
- Original Message -
From: "* KAPIL *" <[EMAIL PROTECTED]>
To:
Sent: Sunday, April 03, 2005 11:52 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] Question
They charge you by the minute for the second and more concurrent calls if
you are on an unlimited plan.
- Original Message -
From: "JD Austin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 04, 2005 1:41 PM
Subject: Re: [Asterisk-U
I have a clean install of rapid 1.1 installed. I have installed the Flash Operator Panel from the Install Other Software menu. I am able to log into the panel from another computer on my network but all I see is the Conference Room 300. There are no extensions or any other options on the panel.
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