DISA doesn't connect the caller to a PSTN dial tone, it just happens to
(in the US at least) sound like the PSTN one. It is asterisk providing
the dial tone, waiting for digits to match to a specific context. Just
make sure that the context they are in, like John said, contains the
Please try to redirect those questions to the appropriate place,
I mean the A2Billing forum : http://forum.asterisk2billing.org
It's off-topics for the Asterisk-user mailing-list.
Kind regards,
/Areski
On 9/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Everything was going well, I got
I was going to use a Dell 1425 for Asterisk build but I see on Digium's
website that hardware may be problematic. Can anyone shed a litle more
light on the problem. I see the Intel ethernet cards seem to cause
problems. If I need to disable the onboard Intel on the Dell hardware I
can I
MF wrote:
Has anyone got a clue about this?I need to know which operator to
send a message to, prior to the queue command ringing him, (just
after he is assigned)
Anyone knows if I can get to know the operator ACD choosed to send
the call by using Realtime Queue, or maybe via the
i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats possible, without tying up a
This seems to happen when an agent makes an attended transfer. Does
anyone have more information?
Michael Welter wrote:
There was activity in late 2005 concerning PRI channel lockups. The
telco sends a call to channel n, but Asterisk thinks channel n is
busy and rejects the call. There
I finally got them to workhave your phones configured like normal.just includes lines like these for mac.cfgprgkey6 type: blfprgkey6 value: 701prgkey6 line: 1prgkey7 type: park
prgkey7 value: asterisk;700prgkey7 line: 1where ... line: 1 is registered on the asterisk server as [phone]then in
How do you remap the Services key?On 9/11/06, Shawn Kelley [EMAIL PROTECTED] wrote:
Hi,
Does anyone know how to do a re-map of a key on the Polycom
to make it dial a number.
I know how to remap a key to a certain function, but I don't
know how to make it dial a number.
I'm wanting
Hi Folks,
I'm getting a lot of these messagse now with the Grandstream phones and Asterisk
Incoming call: Got SIP response 415 Unacceptable Content-Type back
from 192.168.1.X
I don't think I noticed them when I only had one or two phones hooked
up for testing, but I suppose I could have just
I'd like to be able to record a voicemail message, then enter a list of
extensions to forward that voicemail message. Currently, the
sendvoicemail=yes setting in voicemail.conf [general] section only
allows for one extension to forward the message. Is thee any way to
forward to multiple
Hi Guys
My ITSP told me that i need to change the rtp rfc2833 payload from
101 to 97, where and how can i set that???
Then i should change the sdp 200 message from
a=fmtp:97 0-16
to
a=fmtp:97 0-15
i have no idea about it, so... any suggestions, how can i set that???
Thanks in advance
BR
--
o o wrote:
Hoping someone can point me in the right direction. I
have the following setup:
It is worth noting, that if you have a console open and you run out of
licenses, (I don't know at which verbose level this is) You are made
very aware of it.
IIRC roughly 10 messages a second warning
Hi,The answer for your question is Yes for Sure.Check VoiceMail( ) application syntax:VoiceMail([flags][EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]
)http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail
Hope it helps,MOn 9/11/06, Jay Dutt [EMAIL PROTECTED] wrote:
I'd like to be able
What provider?
Pots lines?
SOME providers will provide hunting on residential lines, but not all,
and most probably not 4 lines. Hunting does not require any thing more
than the providers switch programmed to do so, but most will not do more
than two lines.
VOIP it all depends again on the
In the forthcoming 1.4, you can tell the Queue application to run an AGI just
before sending the call to the destination. In the AGI, you can use the (also
new in 1.4) MEMBERINTERFACE channel variable to determine the destination.
Of course, that's not a solution now since 1.4 is not even
Some telcos will set up your lines so that the calls will go to
available lines, rather than the first one.
PaulH
AsteriskIT
On Mon, 2006-09-11 at 16:01 -0700, Christopher Corn wrote:
i plan on buying 4 residential lines for our small office and i was
giving some thought. we'd like to have
I have a couple of clients with a bunch of GXP-2000. They can do
attended transfers with no problems. However, there are times that
the party to transfer to is simply not at their desk and the party
wanting to transfer the call knows that. In these cases, they'd like
to blind transfer the
All of a sudden, it started to work. I tried new and old settings again, tried stun.xten.com too, but now it works even without any new settings. I couldn't understand why was it not logging in before when I had the same settings.
___
--Bandwidth and
--- Thomas Kenyon [EMAIL PROTECTED] wrote:
o o wrote:
Hoping someone can point me in the right
direction. I
have the following setup:
It is worth noting, that if you have a console open
and you run out of
licenses, (I don't know at which verbose level this
is) You are made
very
Both trixbox and asterisk2billing have their own lists... you may have better luck searching there.
bp
On 9/11/06, Steve Totaro [EMAIL PROTECTED] wrote:
Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see
both application is not integrated special on cdr
Christopher Corn wrote:
i plan on buying 4 residential lines for our small office and i was
giving some thought. we'd like to have one main number that can transfer
calls to the other lines. but seeing that i have 4 different individual
lines with different numbers, im not seeing hows thats
rich, thanks for replying. i assume your talking about enabling call forward and call forward on busy from my vsp side. i dont quite grasp everything else that your saying, can you explain in laymen terms. thanks.Rich Adamson [EMAIL PROTECTED] wrote: Christopher Corn wrote: i plan on buying 4
okay, i undrestand what you guys are saying. thanks alot.Brent Franks [EMAIL PROTECTED] wrote: i would think i would need one DID with multiple simultaneous connections.Hello, you can't setup a DID per se on an analog line.Essentially what you want is 4 regular POTS line in a hunt group.The
all thanks for the replies. i know what to do now. thanks.John Novack [EMAIL PROTECTED] wrote: What provider?Pots lines?SOME providers will provide hunting on residential lines, but not all, and most probably not 4 lines. Hunting does not require any thing more than the providers switch
Hi,
I would first of all know
which frequencies Polycom HD Voice use 16 kHz? Also, now that Polycom as
released a phone that support 16Khz sound and that more device
will probably support this in the near future, is there any plan to support higher
frequencies in Asterisk. For sure
im having issues when routing calls from the outside with my new VSP. this is what asterisk tells me when i try to make an incoming call, i get the no service response when i call. -- Executing GotoIf("SIP/christopher_corn-eddb", "1?from-trunk||1") in new stack -- Goto (from-trunk,s,1) --
I'm pretty sure you can't do that. You can map a key to perform a single
function, such as perform the operation of another key, or dial a SINGLE digit,
but you can't make it dial a series of digits.
Doug.
-Original Message-
From: Matt Birmingham [mailto:[EMAIL
Rushowr wrote:
Hugo wrote:
Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic
Realtime successfully. In fact, I want to know how to compos the correct
DB(postgres or mysql) fields (I think STATIC configuration is different
from DYNAMIC).
Regards,
Hugo
Beeing able to be called on 2 different extensions and still get a single missed call list seems very useful for me.I hoped it could be dealt on SIP level addind specific parameters to SIP INVITE messages, for example, but I couldn't find any clue for that.
Maybe, the right way to do it is to ask
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