I have been down this path with Grandstream but they (for reasons I don't
understand) want to upgrade the firmware to have a dial plan.
So the best you can do is use early dial, for all fixed length numbers in
the * dial plan this works reasonably well. International numbers vary in
length so
Record a ring tone file as the default
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Queue option r, like so:
Exten = s,1,Queue(somequeue|r)
Try 'show application queue' at the CLI
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED]
Hi Michael, in practice I think that the managers extension should default
to the assistant who can screen the call or call forward it.
Call Forward - always or Call Forward - no answer would give you the
flexability required.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
The + sign is grammatic only it just means your international dialing prefix
+ the country code etc.
So for dialing a number from Canada to the UK you would advertize the
number as + 44 xx etc. In Canada we dial 011 for international
calls so I would actually dial 01144 xxx
Yes thats the bottom line, its mostly the country code which can be 1-3
digits long. There is no rules based solution for this. Historicaly each
country picked a number out of a hat except the US (which had to be
number 1) because as we all know it's the centre of the universe. The
former USSR had
I would be very interested in getting an 8 port FXO myself. They are very
new so I don't think there are any used ones out there yet.
Does anybody out there in Canada stock them yet?
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Has anyone used either the 8 port or 4
Sounds like you have a disconnect supervision problem.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
We currently have a pri coming into our asterisk system. Most of the
time, the did numbers that we call into it work great. However,
occationally, we get fast busies,
You might want to take a look at the new 4 port FXO from Grandstream
I haven't had one yet to evaluate but assuming it works it is very price
competative and off-loads all the analog (TDM) stuff from your PC
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
I have been using
The Message Waiting Lamp (neon) on these phones requires a 90v signal
which is generated and switched to the phone via a special station card
on an analog PBX. This feature was developed mainly for Hotel and Motels
but I doubt there are any manufacturers who would develop this
functionality for
not want to answer the call, he/she simply hangs up
and you will be back to your original conversation.
The callee is put on hold automatically
Eric, attended transfer is only possible with an ATA??
On 12/5/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Henry.L.Coleman wrote:
Attended
Attended transfer is really four functions
1. Put the caller on Hold while you dial another number
2. Speak to the dialed number (announce the call)
3. Patch the call on hold to the other party using transfer button.
4. Disconnect (otherwise this would be a 3 party conference)
How these functions
This 24/7 mantra that companies keep promoting to us is often just the
ability to subject us to endless hours of their lame MOH while you wait
for the one service specialist to answer the phone from Tinbuckto.
My apologies if you live in Tinbukto.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
-timbuktu.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry.L.Coleman
Sent: Tuesday, December 05, 2006 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] any possibility of Vonage Integration
Hi Scott, I have the following firmware
1.1.0.16
1.1.0.11
1.1.1.9
1.1.1.14
1.1.2.6
1.1.2.13
Some of these were not from the official website but they were all an
improvement 1.1.2.13 is very stable apart from the 56 button ext, unit
support.
Let me know which ones you want and I can send them to
Using the PSTN in Toronto ie 416 NXX X all calls to 647 and 416
exchanges are local. 905 is an over-lapping area code, most excahnges are
local, however Whitby (905 430 ) is Long Distance while 416 428
(Ajax) is not. You can find out which ones are long distance (from the
CRTC web
Hi Nigel,
If I understand your question correctly, you can accomplish what you need
in Trixbox/FreePBX by having your calls answered by a queue. When the
caller is in this queue, he will hear music on hold until the call is
answered by an agent. When the agent answers the call a recorded
I have deployed the Grandstream 2000 with very little hardware problems.
Early firmware was petty rough but from 1.1.1.9 onwards is very robust.
Frankly it represents the best bang for your buck. The only thing that I
would like to see is a dial plan (which would speed up dialing). Most
IP-phones
By the time you purchase PCI cards for you extensions (FSO ports)you would
be better off purchasing SIP phones like Grandstream GXP 2000 this will
give you a fully featured PBX IP phone for about the same cost or less
than FSO ports. Asterisk will have no problem running 25 or more SIP
phones
and the
ability to have extensions in multiple sites/offices without any line
costs.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Henry.L.Coleman wrote:
Its a bit like the VHS vs Beta war, both systems have their good and bad
points In the end, sales/marketing perception
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry.L.Coleman
Sent: Wednesday, November 01, 2006 7:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I came to the same conclusion.
There is one thing
Hi Andrew, I can highly recommend using the Granstream GXP 2000.
Upgrade the firmware to ver. 1.1.1.14 and you won't have any problems.
The 4 line buttons are not actual lines they are calls queued up on an
extension so you can have as many incoming lines as you want. The first
call comes in on
I strongly recommend you upgarde to the latest firmware for the GXP 2000.
I have been using them for almost a year now and while the early firmware
was poor they are now very stable and working fine (from 1.1.1.9) onwards.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the
Obviously we (as an industry) have to start to take notice of this spoofing.
otherwise big brother will start to legistrate against it. This will
give the CRTC or FCC another excuse to spend a lot of tax payers money on
something which is of marginal value.
My position is that there are only two
As I understand it the main advantege IAX has over SIP is the number of
port it uses and therefore its ability to traverse router/switches and
firewalls
Also the higher number of simulatanious SIP calls travelling through these
devices adds a higher overhead than IAX with it's single port.
like the VHS vs Beta war, both systems have their good and bad
points In the end, sales/marketing perception will always win regardless
of better technologies.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
On Thu, 2006-10-26 at 13:14 -0400, Henry.L.Coleman wrote:
As I
You are welcome. Please let me know if this makes any difference.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Henry.L.Coleman wrote:
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
and most loop start interfaces don't really care (they work
I believe you have to buy the non-freeware version to have this enabled.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote:
I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
initiate time, but I can
Just a thought ... try reversing the Tip and Ring
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
You have polarity reversal detection and I do not (I did try with it on,
but it didn't help even though there I have measured a polarity reversal
on disconnect)
FWIW: I
a similar problem with Foriegn Exchange line (FX) but I haven't had
time to visit the client to check this out yet.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Henry.L.Coleman wrote:
Just a thought ... try reversing the Tip and Ring
Henry L.Coleman CEO
Henry
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various tones according to different country standards
would it be possible to
The latest X-lite version has autoanswer button on the front.. marked AA
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Hi Greg,
Idefisk support Auto-answer only in a biz version
I suppose you got free version..
You will find more details
Hi Andrea,
Try the following:
featuredigittimeout=1500 ; Slow down digits for the record
[featuremap]
automon = *0 ; One Touch Record
Use both option switches(wW)
Check that the dial plan on your SIP phones doesn't preclude this feature
code.
Henry L.Coleman CEO
I have a bata site we can use to test your software.
Please contact me [EMAIL PROTECTED]
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Secure multi-tenant partitioning capabilities?
What is your distribution intentions, commercial or GPL?
-Original Message-
The quirk of your old PBX is in fact exactly what happens when you put any
two analog phones on the same line. The easiest way to duplicate this is
to connect another analog phone to your ATA. Some analog phones can
indicate when the other is on the line and can put a call on hold locally.
Henry
The quirk of your old PBX is in fact exactly what happens when you put any
two analog phones on the same line. The easiest way to duplicate this is
to connect another analog phone to your ATA. Some analog phones can
indicate when the other is on the line and can put a call on hold locally.
Henry
I have had this problem before and it always turns out to be the fire wall.
You SIP registration and signaling (port 5060) is going thru okay but the
audio signals use a range of different ports which (if blocked) will cause
the problems you experience. Try putting * in DMZ to test this theory
Frankly waiting for the box to break will loose you the client.
I would change the box but use the original Hard Drive, it only takes a
couple of minutes on a small system.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
On Wednesday 11 October 2006 15:16, Douglas
I can think of a couple of ways to achieve testing of a PSTN line but this
would seem to be the easiest.
Attempt to call an incoming PSTN/SIP/IAX line from your outgoing PSTN
trunk, answer the call at a vmail box and notify you of a message via
email.
insert a delay of x minutes and do it again.
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