2009/7/1 Geraint Lee gera...@gmail.com:
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your own hardware controlled in the
same method as wireless AP's
Hi,
Is anybody picking up emails as attachments on an android phone like
the t-mobile G1?
I had this working a while ago but since I re-installed my asterisk
box to a newer build I am unable to open
the attachments, I just get told it can't handle the format?
I've been through and tried all wav,
2009/6/8 Wai-Sun Chia waisun.c...@gmail.com:
Fellow Asterisk Users,
I'm trying to marry SugarCRM and Asterisk..perhaps starting from elementary
features like a pop-up CRM record upon receipt of inbound call, for
starters.
Anybody who has successfully done this and beyond?
What integration
H, not sure about you but I often pick up a book and flick from
the back to the front, does nobody else do that?
On 05/12/2008, Atis Lezdins [EMAIL PROTECTED] wrote:
On Fri, Dec 5, 2008 at 3:47 PM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
wrote:
Tzafrir Cohen wrote:
Top posting
Hi,sorry for off topic post, struggling to find any information on UMA in
the UK. I have a Blackberry 8320 phone with wi-fi and UMA
capability, its actually an unlocked Orange branded phone.
T-Mobile don't support UMA in the UK, is it possible to do anything else
with the UMA feature of this
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our end reliably.
If we originate the call, we can
On 18/02/2008, srinivas Antarvedi [EMAIL PROTECTED] wrote:
Hello all,
I am struggling with sending voicemail as an attachement in Email.
When i have given the email like [EMAIL PROTECTED] it is delivering
to my gamil account perfectly(of course to spam folder).
But when i given the email
On 07/02/2008, Brent Davidson [EMAIL PROTECTED] wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to eliminate long-distance costs incurred from the
constant branch to branch calls. We're using the Snom 300's at all
offices for the desk phones and
On 23/01/2008, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Has anyone ever seen an Snom320 lose settings?
It's been working fine for months and then I got a call this morning
saying that it was asking for country, timezone etc.
I logged in
Hi,
just wondered if it was the same firmware on both devices?
thanks
Mike
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Hi,just spotted this on the RIM site.
http://na.blackberry.com/eng/services/blackberry_mvs/
Just wondered if there is anybody working on somehow linking MVS and
Asterisk, or if it is even possible?
thanks
Mike
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Hi,
I've had a Snom 300 connected to my Asterisk box at home for 12 months
or so now. Recently it lost all its settings and I had to reconfigure
it via the built in website.
For a few weeks it was fine. Couple of days ago it lost its settings again.
I logged in to its web server and thought I
Hi,
just noticed chan_mobile, which looks like it will do exactly as I need.
http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels
However seems it is only for latest 1.4 but there is a mention of a
backport for 1.2
http://www.sigsegv.cx/sip-9.html
Anybody using this with something like
On 7/6/07, C F [EMAIL PROTECTED] wrote:
Have you tried wav49 format?
Yes, I have
format=wav49|wav
Mike
On 7/6/07, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
I recently upgraded the firmware on my Blackberry 8700 to 4.2, this
seems to give
it the ability to play wav files.
I wondered
On 7/6/07, Dave Bour [EMAIL PROTECTED] wrote:
Can you see an attachment? If so, does it download?
Yes the attachment is there but it seem that it will not download it,
which leads me
to believe it does not understand the format?
thanks,
Mike
Dave Bour
Desktop Solution Center
an option to down as a result. Do you have
the same?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent: Saturday, July 07, 2007 4:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT
itself doesn't register an
attachment (ie, if on the main email screen, no paperclip on the
envelope). It doesn't give an option to down as a result. Do you have
the same?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent
Hi,
I recently upgraded the firmware on my Blackberry 8700 to 4.2, this
seems to give
it the ability to play wav files.
I wondered if anybody out there had managed to get their BB to play
the wav files as
attached to the Asterisk voicemail emails?
Mine seems to ignore the attachment.
I am using
Hi,
would it be possible to use Asterisk to record calls only? There would
be an existing PBX and calls come in on a ISDN30 line?
The Asterisk box would need to sit between the incoming ISDN 30
circuit and the existing PBX.
Is this possible?
thanks
Mike
Hi,
is there a way or feature available in Asterisk where one can 'pull' a
call back from
voicemail.
i.e. if you don't get to the phone in time and it goes to voicemail,
can you key some
sequence in and pull the caller out of voicemail and speak to them?
Thanks
Mike
On 5/2/07, Tim Koehler [EMAIL PROTECTED] wrote:
Hi,
I can agree for smaller installation/home offices the Linksys WRT series is
pretty good (I'm using this at home).
I'm using the dd-wrt Firmware (www.dd-wrt.com ) which is also available for
plenty other routers.
With QoS values set right I
Hi,
Just wondering about using Asterisk for hosting a support line with a few
distributed operators taking calls.
My idea was for the calling party to be prompted for their support
number (maybe a unique
4 digit number) this would then get looked up in a database and then
place a pop up on
the
On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote:
I also restarted the box and the problem is not solved :(
PS
Pedro,
I had the same problem on a test box recently. I fixed it by upgrading
the version
of FreePBX to 2.1.3.
Mike
2006/11/18, Dumpolid Exeplish [EMAIL PROTECTED]:
i also used
On 11/9/06, Steve Totaro [EMAIL PROTECTED] wrote:
Martin Joseph wrote:
On 2006-11-08 14:40:09 -0800, Ken Williams
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
After about one weeks time I've gone from no VoIP to a completely
configured system for two of our
On 10/31/06, Joao Pereira [EMAIL PROTECTED] wrote:
Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need
to focus more in SIP and Asterisk compatibility and less in pricing
Hi,
I've been asked to put together a quote for a system which basically
will be a virtual PBX based on Asterisk with some IVR's and a whole
bunch of GotoIfTime's.
There will be one incoming DID via SIP, user gets dropped in to an IVR
and then depending on the option they choose, *and* the time
Hi,
I wonder if anybody can share their experience of this. I am designing
a system with 90
GotoIfTime conditions to check through for a match.
Bascially each month day will be split in to 3 time ranges and a month
has 31 days in, so this gives a possible 90+ combinations, nearer the
end of the
On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can anyone direct me to a colo provider in the UK where I can park an
asterisk server and buy UK toll free inbound services over SIP?
Thanks
Probably more relevant on the asterisk-biz list. However I'd be
interested to know what replies
Hi,
I've recently got a Snom 300 phone.
When I set it up and a VM was left the light flashed as excpected.
I can use the soft button with the Tick sign on it to go straight to
voicemail, all fine so far.
However once the message is picked up and listened to, the light still flashes?
I'm using
On 9/25/06, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
I've recently got a Snom 300 phone.
When I set it up and a VM was left the light flashed as excpected.
I can use the soft button with the Tick sign on it to go straight to
voicemail, all fine so far.
However once the message is picked up
On 9/25/06, yrving rivas [EMAIL PROTECTED] wrote:
I would like to know if any of you have a cell phone like a pci card to
install in one slot to my asterisk server?, I want to make a connection from
my asterisk to the cellular network.
Does anybody has a solution like this?
Regards,
Yrving
When you installed Trixbox did you not boot from the Trixbox install CD?
This installs CentOS and Trixbox.
I'm curious how you installed Trixbox?
Mike
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc.
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
sorry i made a mistake telling you that i installed it using rpm package.
actually there is no rpm package for trixbox to download. you have to
install it using .tar.gz package using ur existing linux OS. so sorry about
that.
Yeah thats your
On 8/9/06, Stephen G [EMAIL PROTECTED] wrote:
Hi there,
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
Hi Stephen,
I have an A200 with 1 x FXO module board in it.
I'm in the Uk using a BT line and am having a lot of
I'm having real problems getting my Sangoma A200 card with FXO board in to
detect hangup at all.
Basically if the remote end hangs up the call, Asterisk does not seem
to detect a
hangup.
A month or so ago I was running a system with 2 x X101P cards in, this
detected hangup fine.
Since switching
On 7/6/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On Jul 6, 2006, at 6:01 AM, Peder @ NetworkOblivion wrote:
Is there a buddies feature on the Cisco phones, like there is on
the Polycom? If not, how are people getting around the issue where
a receptionist wants to see who is on the
Hi,
I've recently been testing an A200 from Sangoma with a single FXO card
(2 ports).
I'm pleased with the card apart from hangup, or lack of it.
I'm in the UK, so I am wondering:-
a) Anybody else in the Uk using the same card on BT lines
b) which method of hangup detection is used on BT
On 8/3/06, Khaled Chehab [EMAIL PROTECTED] wrote:
I have trixbox 1.0 how I can update it to 1.1 or from where I can download
trixbox 1.1
Have you tried the trixbox-update.sh script?
Mike
Regards
*
No
Just wondered if you fixed your hangup problem?
Mike
On 6/29/06, El Flynn [EMAIL PROTECTED] wrote:
chan (Alpha Trilogies Networks) wrote:
Hi,
Does some one experience the Sangoma A20X-ec series card that cant detect
the hangup tone?
snip
[channels]
context = from-pstn3
switchtype =
Hi,
can anybody confirm if there are any patches required for Caller ID to
work on a Sangoma A200 card on a BT line in the UK?
With Asterisk 1.2.9.1
thanks
Mike
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Hi,
I'm interested in anybody that is providing a phone support service
using an Asterisk system, with built in charging system.
I run a PC support company and use Asterisk at the home/office. I
would like to be able to provide technical support to my customers
using asterisk. However I want to
On 4/13/06, David Cook [EMAIL PROTECTED] wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely cool.
On 4/3/06, Steve Jones [EMAIL PROTECTED] wrote:
I've been using my Asterisk (At my house - 2 modem-type fxos, and an
assortment of SIP endpoints for phones) for about 5 weeks now, and I've been
really happy with it, but I'm still having an echo problem that I've
exhausted google with, and
Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?
thanks
Mike
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On 3/22/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
Hadley Rich wrote:
Hi all,
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
I couldn't make TDM400/FXO work on my 1.2.5
On 1/17/06, steve [EMAIL PROTECTED] wrote:
Message: 20
Date: Mon, 16 Jan 2006 00:18:18 +
From: Mike Hemstock [EMAIL PROTECTED]
Subject: [Asterisk-Users] Choosing an FXO card
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
: 0 XT-PIC acpi
10: 23600597 XT-PIC eth0
11: 513008566 XT-PIC wcfxo
14:6166621 XT-PIC ide0
15:482 XT-PIC ide1
NMI: 0
ERR: 0
Look for the wcfxo's.
Mike
- Original Message -
From: Mike Dent
I'm getting much nearer in getting my Sangoma A2022-SO analog card working
with Asterisk 1.2, however I am unsure of the ordering of ports on the
rear of the card.
I've taken some pictures of the card in the hope anyone can help me
guess which physical ports relate to the 2 x fxs and 2 x fxo
On 12/2/05, Jess Coburn [EMAIL PROTECTED] wrote:
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1 cards
together. I realize asteriskathome is traditionally used for at home, but
I'd like to use it in a small office with a T1 and our hardware is a Sangoma
card. I know
On 12/2/05, Jess Coburn [EMAIL PROTECTED] wrote:
I didn't see it there last time I loaded [EMAIL PROTECTED] but maybe something
has changed recently, this was about 2 months ago. I'm probably going to
attempt to tackle it this weekend.
On 12/2/05, Mike Dent [EMAIL PROTECTED] wrote
Hi,
I'm testing a Sangoma analogue card with two modules cards installed
(2 x FXO and 2 x FXS total)
I'm actually doing the testing so far in a new install of [EMAIL PROTECTED] 2.1.
The kernel is seeing the card but I'm struggling with my understanding of the
zaptel.conf file and possibly the
On 11/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Without putty, my windows would be meaningless.
PaulH
Subtle Paul! but nice! :)
Mike
UK
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Kristian,
Excellent thanks..
On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Hello Everyone,
I have finished up work on what will (hopefully) become AstLinux
0.3.0.
AstLinux 0.2.9 has been released as a test release, and includes the
following changes:
-
On 10/31/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Is there a way to add timestamps to each line in the console so you know
exactly how long a call took? Or is there another way of telling directly
within the console?
I must say its something I would really like to see on the console
suspended.
Anyway I contacted them, lets see what happens.
Mike
On 7/17/05, S. William Schulz [EMAIL PROTECTED] wrote:
Mike Dent wrote:
-- Called NBhX:[EMAIL PROTECTED]/12124565900
-- Call accepted by 66.234.228.160 (format ulaw)
-- Format for call is ulaw
-- Hungup
Hi,
Sipgate works, I can dial out fine. When I make a SIP call with
sipgate (to pstn) I am presented with the ring tone (US style) from *.
If the remote end answers at the start of one of these ring cycles
from asterisk, it does not
interrupt the ring and the remote end just hears silence until
Hi,
I just setup a VoipJet test account (one with 25c credit) to test,
they seem to offer
good rates to 02 Uk mobiles :)
Anyway, everything went ok, iax.conf amended and extensions.conf too,
however when I
try to make a call I see:-
rt*CLI
-- Executing SetCallerID(SIP/2008-d747, 4153574000)
Hi,
for some weeks now I have been unable to make calls via my voicepulse
connect IAX account?
When I attempt the console looks like this:-
rt*CLI
-- Executing Dial(SIP/2008-cf55,
IAX2/NBhXX:[EMAIL PROTECTED]/12124565900) in new
stack
-- Called NBhX:[EMAIL PROTECTED]/12124565900
Hi,
I've had an inquiry for a small UK call centre, mostly outbound calls.
I get the impression they
are mainly calling 3G mobile phones, monthly phone bill, with calls is
approx £5,000 for several
feature lines.
How feasible is something like this with asterisk?
I guess one big question is
On 6/30/05, Mark Charlton [EMAIL PROTECTED] wrote:
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
Sorry I should have paid more attention to your post :)
You are already using fastsms!
oops.
;exten = h,1,HasNewVoiceMail(30${dialed_extn})
;exten = h,2,goto(h,100)
;exten = h,102,DeadAGI(fastsms|44000|Caller ${CALLERID} left
a new voice
mail at ${DATETIME} on Sales extn
Hi,
I'm wondering firstly if somebody already did this and what the best
way might be to go about it.
Basically I want to have a context in * which could just be a regular
extension that you can transfer an incoming caller to.
This extension could then do several things, the one bit I need
Hi Dean
On 6/23/05, Dean Collins [EMAIL PROTECTED] wrote:
Great idea but can I ask how you intend to do the location/destination
entry?
Ahh, I dont think I explained my idea clearly. The directions which
would get sent to the mobile phone are just static/fixed directions.
ie you
Samy,
Sorry I'm late. How about a shameless plug for my distro - AstLinux.
It sounds like it may work well for your needs:
http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3
--
Kristian Kielhofner
Hi,
I like the look of this, however I'm not ready to go the
Hi,
just suppose you have a small LAN in an office with an external ADSL connection.
Is there a problem by dual homing the Asterisk box (one interface on
the local LAN and
the other on the ADSL side). Then making all local traffic heading off
site to go via
the Asterisk box. On this box you run
On 5/19/05, Steve Totaro [EMAIL PROTECTED] wrote:
Does anyone know the best way to automate the deletion of monitor files
after they age two months?
Thanks,
Steve
___
Something like:
find /files/to/check/ -mtime 60 -exec rm {} \;
put
Hi,
it seems if a user leaves voicemail and hangs up the call when done, then
HasNewVoicemail never gets called on the next line in the context.
However if they press # to finish their VM, then it moves to
HasNewVoicemail and this
works?
eg:-
..
exten = 2002,3,VoiceMail(u${OFFICEVM})
exten
Hi,
Is it possible to put some kind of bridge which will do traffic
shaping/prioritising between
my 6 external IP addresses and my PPPoA modem interface?
My other option is to put some kind of device at the edge of all my
networks to shape the
traffic in/out. I'd rather do it in one box if
On 5/16/05, Steve Kennedy [EMAIL PROTECTED] wrote:
On Mon, May 16, 2005 at 03:51:28PM +0100, Mike Dent wrote:
Hi,
I'd be interested in comments from any users of the vonage service in the
UK?
http://www.vonage.co.uk is the website.
Where are the servers located, traceroute would
On 5/17/05, Brian Roy [EMAIL PROTECTED] wrote:
On 5/16/05, Nathan Pralle [EMAIL PROTECTED] wrote:
Hi all.
I'm curious to hear about other people's HOME usage of Asterisk. Do you
have a really neat setup for home use? Fun stuff with VM and/or
forwarding and custom scripts?
If you
Hi,
I'd be interested in comments from any users of the vonage service in the UK?
http://www.vonage.co.uk is the website.
Where are the servers located, traceroute would be useful.
What is the general reliability like?
Thanks
Mike
___
Asterisk-Users
I wa shaving similar problem.
I had to do a proper stop and start of asterisk and now it works!?
Mike
On 5/7/05, Sahil Gupta [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED]:~# mpg123 -v
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and
gwiaxt01.voicepulse.com working fine 5 mins ago, just used it.
Mike
On 5/11/05, Trevor Harrison [EMAIL PROTECTED] wrote:
Anyone else using Voicepulse? This morning I noticed that they seem
to be doa... no dns resolution, no ping, etc.
-Trevor
On 5/2/05, Gary Stimson [EMAIL PROTECTED] wrote:
On Friday 29 April 2005 23:20, Paul Tyreman wrote:
What are you using instead of SIPGATE in the UK ?
I also have this problem with DTMF tones not being passed to Asterisk from
a PSTN line and my e-mails are being ignored too !
Both are
On 5/9/05, Areski [EMAIL PROTECTED] wrote:
Dear All,
Here the version 2.2 a new version of your dear CallingCard Software !!!
http://www.areski.net/areskicc-doc-v2/
Many new features have been added and several enhancements made!
HI,
looks good! Any chance of it working with MySql do you
On 4/30/05, Jason Brown [EMAIL PROTECTED] wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind
Hi,
I notice when you call digium and choose the option for tech support,
it asks you to enter
your reference number, it then looks up the job?
Does anybody know what they are using to do this?
I'd like to do this with asterisk and Request Tracker, maybe digium
are already doing somehting
Hi,
is anybody using Asterisk to input tickets to Request Tracker?
I tried an older version of this but it did not seem to work with the
new version
of RT I have running.
I think it would be excellent to be able to call a helpdesk and then leave a
message which is then attatched to an RT ticket
Hi,
for some reason my Asterisk has been crashing recently. I've not made
any changes to
Asterisk itself, although I did attempt to get the asterisk RT
gateway working a few days
ago, which required me to install an AGI file (now removed) and also
install Asterisk-Perl?
Anyway, I'm getting dump
-
From: Mike Dent [mailto:[EMAIL PROTECTED]
Sent: Friday, April 08, 2005 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk and RT (Request Tracker) setup?
Hi,
I'm attempting to get Asterisk and RT working, I've followed
Hi,
I'm attempting to get Asterisk and RT working, I've followed the
details from http://megaglobal.net/docs/asterisk/html/rtasterisk.html
When I dial the extension I put the rt agi on for testing all I get is
silence and on the * console
I see:
rt*CLI
-- Executing Answer(SIP/2010-f6f5, ) in
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
thanks
Mike
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On Mon, 14 Mar 2005 14:42:00 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Mon, 2005-03-14 at 17:31 -0300, César Davi Ávila do Nascimento wrote:
Talk about skype is forbidden, but to be impolite is allowed...
Great list!
Did you ask about skype in asterisk? Did you ask an asterisk
On Mon, 07 Mar 2005 21:32:29 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
The Sipura 841 goes for under $90. It works well, and has a nice web
interface. Once you get more advanced you can use their Sipura Profile
Compiler tools to work with multiple phones (or work through
I have A cisco 7960 and couple of Budget Tone's. I'm hearing more echo
on the BT's than on the Cisco, this is with AGGRESSIVE_SUPPRESSOR
defined
and calls coming in via my analouge lines to X101P clones (in the UK).
The Cisco echo/choppiness is not nearly as bad as the BT.
Is it likely some
On Tue, 8 Mar 2005 07:45:42 -0800 (PST), Victoria Alexandru
[EMAIL PROTECTED] wrote:
I'm a neewbie in Linux, so please bear with me.
I have a school assignment to make communication between 10 SIP softphones
(kphone).
So far I got trouble installing Asterisk. The information in asterisk web
On Tue, 8 Mar 2005 16:55:30 +0100, Luca Bariani
[EMAIL PROTECTED] wrote:
Hi
I'm just subscribed to this list because I'd like to try Asterisk
I'd like some soggestions for getting started with it
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think
Works sweet here with a 7960G too, 7.3 SIP fware.
Mike
On Mon, 7 Mar 2005 11:41:38 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?
We have just about tried
Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike
On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
No, I dont mind paying more for something if I know its going to be reliable.
On Sun, 6 Mar 2005 16:36:16 -0700, The Phone Guys [EMAIL PROTECTED] wrote:
So you want it 100% perfect and you want it for peanuts.
Makes you wonder how many *really* reliable VoIP providers there are out
there?
Maybe at some stage in the future the big telcos will provide VoIP
termination, DID's etc. They may as well make some money from it, I'm
sure they could get it right?
BT providing IAX2 and SIP termination? Hmmm, maybe one day.
Mike
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Asterisk-Users
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCoptm
There is a difference when you are that childs father or mother but
I'm running two in a box Nigel and they work ok for me. Maybe a faulty
card you have?
Is the extension/socket you plug it in to working ok for a phone?
Mike
On Fri, 4 Mar 2005 15:48:53 -, Nigel Taylor
[EMAIL PROTECTED] wrote:
Hi
A quicky has anyone had success with an
On Wed, 23 Feb 2005 13:25:55 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
hehe, maybe I should go ahead and drive myself home and get away from a
computer for the rest of the day. If you are right, I will probably end
Heheh, maybe not just a day!? :)
up totally pissed before the day is
The X101P works but I dont think it would be acceptable in a
commercial environment. The audio levels are too low and there is too
much echo (or speech break-up with the aggressive cancellation set
on).
Saying that hang-up detection works and CLID works with some source
code changes.
Anybody got
Phone numbers beginning with a '1'? Surely not, they should all start
with a 0 :)
Mike
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Hi,
I'm just wondering how other people cost/charge for * boxes they have installed
in small businesses?
Asterisk is such a complex beast and is capable of so much it seems
equally complex to figure out a charging model!?
I'm contemplating installing a pair of servers at a small company with
two
Hi Paul,
I have the Cisco 7960 which has 6 lines. You can configure the other lines with
different extension numbers if you wish and then set your dialplan so that
calls coming in over different circuits, FWD, POTS etc come in different
buttons on your phone.
I'm sure there are lots of other uses
Jeff,
does any email get sent out?
Mike
On Thu, 27 Jan 2005 18:42:25 -0500, Jeff R Glassman
[EMAIL PROTECTED] wrote:
I am running [EMAIL PROTECTED]
Voicemail works fine but does not email out the voicemail attachments. Any
suggestion?
---
Voicemail.conf
or maybe country? or should that be County? :)
Mike
On Sun, 30 Jan 2005 16:49:29 -0500, Tim Mattison [EMAIL PROTECTED] wrote:
Depending on your state. :P
On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote:
Don't forget to warn your callers about the recording.
Tim Mattison
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