Hi Mike,
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
HTH,
Mat
-Original Message-
From: [EMAIL
On Friday 22 September 2006 13:36, Mat Stace wrote:
It's a while since I did this one myself, but I was doing the exact
same thing when using voipbuster (or whichever of it's sisters
services I was using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1
Hi everyone,
I'm in the very early stages of rolling out an asterisk box at work, and one
of the things I'm setting up is a trap for telemarketers ;)
What I want to do is have a sipgate number in the UK here which rings for 10
seconds without calling a hard or softphone, then goes to a
-Original Message-
Adam Dobrin
Bob Goddard wrote:
On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote:
before I accuse somebody to overbill I would like you to
calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for
I currently have 4 lines on my Cisco 7960G, between these 4 lines there are
3 mail boxes (one work, one personal, and two testing lines sharing a
mailbox).
It's not so much multiple MWI as multiple lines with their own MWI, but it
does the job
Cheers
M
-Original Message-
From:
sip show registry ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald_Wiplinger
Sent: 09 August 2005 09:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Forbidden - wrong password on
authentication
Greetings list,
I've been bashing my head against a brick wall for a couple of weeks now to
try and get this sorted, have been scouring google/the asterisk-users list
archives to no avail.
The problem I am having is that one extension (an off-site iaxy) cannot
transfer incoming calls from our
Title: Message
I'm
not exactly sure on the /how/ * mathes items from the sip.conf (I suspect it
goes to the latter for whichever provider), but the way configured my
extenions.conf to handle multiple incoming accounts from sipgate is like this
(obviously much simplified for ease of
Title: Message
That won't help either. Context is
always 'default', but what I want is a different context on any number. Maybe
oej'speermatch branch solves the problem. But I cannot compile it, There are lots
of ' merge right' tags in
chan_sip.c.
How about a slight modification of my
Hi Dean,
In the voicemail.conf, in the [general] section near the top, I've got
; Who the e-mail notification should appear to come from
[EMAIL PROTECTED]
My e-mails now come from [EMAIL PROTECTED], making to easy to set up a
filter in my e-mail client to move voicemail messages into a specific
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mat Stace
Sent: 28 July 2006 14:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Change the from@ using the
voicemail.conf
Hi Dean,
In the voicemail.conf, in the [general] section near
Evening everyone (obviously depends on when you're readin this, but hey).
I'm trying to set up a multi * server situation, and am falling over at the
second server, and after a day of google etc, have come up against somewhat
of a brick wall.
I can make calls each way between the two servers no
thing ;-)
Mat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mat Stace
Sent: 06 November 2006 17:42
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk servers being greedy and
not letting goof the media path. (using IAX2
As of 22:45 GMT it's working for me
Jerry Glomph Black wrote:
This service has been working well lately, but as of this morning is
promptly blowing off IAX connections with the dreaded 'No Authority
Found' error.
Any concrete info greatly appreciated!
Dr G
Lifted from the developer page of the google talk site
(http://www.google.com/talk/developer.html)
5. What protocols are used for voice calls?
Google Talk supports a custom XMPP-based signaling protocol and peer-to-peer
communication mechanism. We will fully document this protocol. In the near
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK landline numbers :)
In my iax.conf
[voipbuster]
type=peer
host= 213.61.187.150
secret=YOURPASSWORD
notransfer=yes
context=default
In My extensions.conf:
exten = _770[12].,1,SetCallerID(CID
account is empty, but when 1euro is deposited, client still works, but asterisk does not. Did you have any problems?
Rudolf
Mat Stace, Colewood [EMAIL PROTECTED] wrote:
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK
Hi list, I'm hoping that I'm being stupid, and someone can tell me
what's going on, but for the life of me I can't figure it out. (it's
been a long day, and I'm now in the last 3 weeks of organising my
wedding, so I hope this makes sense ;) )
When at my desk, accessing (for example) my
Just to answer my own query, I needed to set the devices to dtmfmode=inband
in my sip.conf, and on the 7960 set Sip configuration - Out of Band DTMF -
none
The benefits of a good nights sleep :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mat
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