On Tue, 17 Jul 2007, [EMAIL PROTECTED] wrote:
Hello guys,
Does anyone has an Asterisk server hosted off-site ? Like in those data
centers that do web hosting in dedicated servers ?
Is there a hosting company that has a special plan to host voip services
like this, or usually is hosted in
in a better position to
do this than I am, as you have the phones and I don't!) You'd normally not
need to do anything to the features.conf file to make phone transfers work
using the phone features.
Gordon
Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 18 Jul 2007, satish
patel wrote
On Fri, 20 Jul 2007, Martin Smith wrote:
I'd bet the emails are addressed to the list and the original sender,
both, so for the original person they appear twice, but everyone on the
list gets them a single time. I haven't seen any duplicates.
I've seen list duplicated and sometimes
On Tue, 31 Jul 2007, Yann JOUANIN wrote:
Hi all,
There is a patch in the Mantis that provides a function to reboot GS phones.
In order to use it, I need to know how the digest for SIP Notify messages is
calculated.
If anyone knows that, please let me now
I don't know about the SIP
On Tue, 31 Jul 2007, Jeng Yu wrote:
Hi All,
I have a telephony project for which I need
to build a prototype to demo for management.
The prototype must work on a GSM phone network.
In the demo system, a call from GSM phone comes
into the demo box. The demo box runs CallWeaver.
On Thu, 2 Aug 2007, Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
You really ought to qualify this with the country and the relevant laws
that you think
On Thu, 2 Aug 2007, Rizwan Hisham wrote:
hi again.well i have been trying to know what is the relationship
between asterisk and stun. what i mean is, i understand that a client
requests stun server to know whether its behind a nat or not. if its not,
then its ok. if it is behind nat,
On Thu, 2 Aug 2007, Steve Totaro wrote:
Gordon Henderson wrote:
On Thu, 2 Aug 2007, Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
You really ought
On Sun, 5 Aug 2007, Rory Campbell-Lange wrote:
In the O'Reilly Asterisk book it suggests that it is important to allow
BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that
I don't think. Is this an issue with the Sangoma card?
Probably not. Once the system is built, have a
On Tue, 7 Aug 2007, Olivier wrote:
So no proper logoff between logins, right ?
As I will apply free sitting in school environment, chances are phones would
then remain logged-in several hours or days between another user logs in.
My thoughts are focused on finding the right balance between
On Tue, 7 Aug 2007, Olivier wrote:
Gordon,
What you described is exactly Follow-me feature : users are always logged
and can be reached somewhere.
I've heard of some variants of this feature - that's the beauty (and
down-side!) of a programmable system - it's open to different people's
On Tue, 7 Aug 2007, Olivier wrote:
2007/8/7, Gordon Henderson [EMAIL PROTECTED]:
On Tue, 7 Aug 2007, Olivier wrote:
Gordon,
What you described is exactly Follow-me feature : users are always
logged
and can be reached somewhere.
I've heard of some variants of this feature - that's
On Thu, 9 Aug 2007, Gunnar Schaller wrote:
Hello,
I want to enable call forwarding at my telco. In Germany you can press
*21*destination# and all calls will be redirected to the destination
without interaction with any equipment on my side.
How to dial this with Asterisk and Zap-Channels? It
On Thu, 9 Aug 2007, Paul wrote:
I have the same debian and asterisk version combo running in more than
one location. Some are T1 and some are in data centers. There have been
times when I got such messages and some simple ping/traceroute testing
showed obvious problems at my end or the
On Thu, 9 Aug 2007, MOSBAH ABDELKADER wrote:
Hello,
I want to create a VPN between two Asterisk servers using OpenVPN.
How to configure Asterisk and OpenVPN to do that.
If it's purely between 2 Linux boxes, then you might want to look into
using the TUN/TAP interfaces and running vtund
On Thu, 9 Aug 2007, Steve Totaro wrote:
John Meksavan wrote:
Asterisk Users,
The more I work with the Asterisk 1.2.13 on the Debian Etch, the more
realize there is no real reliable SIP provider. Having two Sip
Providers is smartest thing to do, one being your main provider, while
the
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single
On Fri, 10 Aug 2007, Cesc Santa wrote:
inline ...
On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 10 Aug 2007, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
Installed from binary or compiled by yourself?
I compiled it myself ...
OK, Great.
I just took a peak
On Fri, 10 Aug 2007, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
Installed from binary or compiled by yourself?
I just took a peak at the command: show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do I need a license or compile
On Fri, 10 Aug 2007, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
And BT's 21cn (21st Century Network) is touted
On Fri, 10 Aug 2007, Olivier wrote:
hello,
I would to define and unattended process to configure devices which are
http-server-enabled, use DHCP but do not use TFTP-DCHP to configure
themselves during boot.
Has anyone worked on such subject ?
I was thinking of something like :
On Sun, 12 Aug 2007, Dean Collins wrote:
Powerful enough to run a small asterisk server though not sure if the
drop down in size from a mini-atx or a micro-atx but I'm sure someone
will try.
http://www.geek.com/first-look-via-px1-pico-itx-motherboard
That's pretty impressive!
I'm sure
On Sun, 12 Aug 2007, Dermot Bradley wrote:
I'm sure it will run linux+asterisk just fine with a 1GHz Via C7
processor as there are many platforms out there using that
combination (or the Via C3)
I recently gave up trying to use Jetway J7F2 motherboards (VIA C7 and
VIA IDE/SATA/Ethernet) as
I'm wondering if anyone has seen (heard!) this before. I have a site which
has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice
and I was quite angry when I heard they'd been installed )-: They have an
asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to
On Tue, 14 Aug 2007, Russell Handorf wrote:
Hello all,
I've been asked to look into my home dial plan to see if I can improve
it by an important customer (my wife).
What we would like to have happen is that an inbound call rings all the
phones (This is done). Once one phone picks up, of
On Tue, 14 Aug 2007, Anthony Francis wrote:
looks broken, is there an apps dir in the source directory?
Built OK for me:
unicorn*CLI show version
Asterisk 1.2.24 built by root @ unicorn on a i686 running Linux on 2007-08-11
08:22:22 UTC
I didn't do anything special...
Gordon
On Tue, 14 Aug 2007, Nicholas Blasgen wrote:
I've heard about this, but I really can't seem to find anything on it. I've
got a strange setup that exists only because of firewall issues, and
everything about it seems fine. The setup:
SIP clients - Asterisk (office) - IAX - Asterisk
On Wed, 15 Aug 2007, Matthew Harrell wrote:
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
FWIW: I do something similar, but purely in dial-plan using the astdb -
On Wed, 15 Aug 2007, Nicholas Blasgen wrote:
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten = *99,n,Set(bottles=99)
exten = *99,n(loop),Noop(There are ${bottles} bottles of beer on the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,Noop(Take one done
On Thu, 16 Aug 2007, Diego Iastrubni wrote:
DUD! THIS KICKS ASS!
(I know I am getting into trouble, but hey! it's already in our PBX!)
Heh... Well I updated it and added some lyrics (and the guys from the
website have said they'd put it up!) So if you want to hear a (rather
odd!) mix of me
On Thu, 16 Aug 2007, Nhadie Ramos wrote:
Hi All,
is it possible to choose outbound route by checking the extension of the
caller?
e.g extension that starts with 3 goes to outbound route 1 extension that
starts with 4 goes to outbound route 2. Basically, i'm hosting two(2)
office,
On Thu, 16 Aug 2007, Bill Andersen wrote:
I'm a network admin that maintains 3 commercial Asterisk
servers for my employer.
I am wanting to move away from the pre-packaged commercial PBXs
to a more pure asterisk setup. The systems I have utilize a nice
web GUI to make changes, but it
On Thu, 16 Aug 2007, Bill Andersen wrote:
Gordon Henderson wrote:
I started with (a).
But since you have a dial-plan that does most of what you want, why not
extract the dialplan (extensions.conf, etc.) and start with that?
I may be showing my ignorance here, but from what I 'understand
On Thu, 16 Aug 2007, Bill Andersen wrote:
OK, I understand that. But if I gotta learn how to support
myself to do advanced features, why pay them at all? I'll
just become my own expert :()
That's how I started...
Sit-down and work out what features you want - and do you want them
On Fri, 17 Aug 2007, Andres Jimenez wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to
On Fri, 17 Aug 2007, Doug Lytle wrote:
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n
On Fri, 17 Aug 2007, Hans Feringa wrote:
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway
On Fri, 17 Aug 2007, Hans Feringa wrote:
Thanks for your response. My answers below.
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
My only other suggestion would be to remove the timeout...
Other than that there's no major difference between my setup and yours.
For outbound dialling, I use:
On Fri, 17 Aug 2007, Andres Paglayan wrote:
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send
On Sat, 18 Aug 2007, ram wrote:
Hi
iam using Asteriks 1.2.17
Server Side ( provider Side g729)
clients side gsm
when iam calling, iam getting lot of errors like below
and lot of voice breaks
Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec
gsm. Use RFC2833
On Sat, 18 Aug 2007, Joe acquisto wrote:
There was a discussion a while back about how to pass Calller ID, when
forwarding, as either the calling number, or the forwarding number.
Had something to do with scams IIRC, but could not find in browsing
the archives.
So, is it in the docs?
On Sat, 18 Aug 2007, voiplist wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
Missing $ before the [
Gordon
On Sun, 19 Aug 2007, G B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following
On Sun, 19 Aug 2007, G B wrote:
Hi Gordon,
I did everything that you suggested, however, the symptoms remain.
I set the rtp.conf to use ports 1 to 2
I assured that my router was forwarding these ports. However, the Media
Description Section of the SIP/SD packet (captured with
On Mon, 20 Aug 2007, bilal ghayyad wrote:
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the
On Mon, 20 Aug 2007, bilal ghayyad wrote:
Dear Gordon;
Thanks a lot for your email.
I need one more tracing tool, how can I know the used
port of the IAX on teh Asterisk and wethor the
listening on that port is successully done (ready to
receive on that port)?
Use
netstat -lnveep
to
On Tue, 21 Aug 2007, Vidura Senadeera wrote:
Dear All,
I would like to get community's feedback with regard to RAID1 ( Software or
Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Asterisk 1.2.19
Libpri/zaptel latest release
2.8
On Mon, 20 Aug 2007, Steve Totaro wrote:
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You can use it as an extension(FXS) or a phone line
(FXO). I believe you
On Tue, 21 Aug 2007, Steve Totaro wrote:
I thought that was what the flashing LEDs on the front of the server's
HDs were for (besides showing activity). Some I have seen also have an
LED near the power button to indicate HD problems.
I guess if you are building your own boxen and not using
On Tue, 21 Aug 2007, Bill Andersen wrote:
When I use the CLI (asterisk -r) I get all sorts of info
scrolling past about current activity such as...
-- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack
-- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in
new
On Tue, 21 Aug 2007, David Gomillion wrote:
Now, you can address Asterisk by saying, Computer, raise lights 20% and
impress all of your trekkie friends when the lights turn up.
Sorry - it's gotta be: [1]
Zen, lights up.
boing Confirm.
But I guess not many leftpondians might appreciate
On Wed, 22 Aug 2007, Steven wrote:
For RAID1, I am not sure.
But for RAID 5, You should always use hardware RAID.
If you use software RAID and your CPU spikes for too long, you can
corrupt your disks. I have seen this several times.
Please report this to the linux-raid mailling list,
peers
iax2 show peers
Gordon
Regards
Bilal
--- Gordon Henderson [EMAIL PROTECTED]
wrote:
On Mon, 20 Aug 2007, bilal ghayyad wrote:
Dear Gordon;
Thanks a lot for your email.
I need one more tracing tool, how can I know the
used
port of the IAX on teh Asterisk and wethor the
listening
On Sat, 1 Sep 2007, Jay R. Ashworth wrote:
On Sun, Sep 02, 2007 at 04:38:19AM +0300, Tzafrir Cohen wrote:
You mentioned that the two disks are identical. Hence there's a large
chance that they're from the same batch. This increases the chance of
them failing together :-p
In practice,
On Thu, 6 Sep 2007, Jeremy P wrote:
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state hardware to run
On Mon, 10 Sep 2007, Adrian Marsh wrote:
Hi All,
Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see Got SIP response 405 Method Not Allowed back from
192.168.3.64 but the phone seems to work ok.
Any ideas where it falls over in the SIP protocol? I've
On Tue, 11 Sep 2007, Juan Sandro wrote:
Hi
We have a number offices accommodating 4-6 people each hence it is very
important for PBX to be fanless and silent. We have been looking at using
IDE flash disks also called DOM. The performance tests we have done so far
satisfy our requirements,
On Wed, 12 Sep 2007, Clayton Milos wrote:
- Original Message -
From: Phil Reynolds [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 8:57 AM
Subject: [asterisk-users] Generating an old-fashioned dialtone
Is there a way to generate an
On Wed, 12 Sep 2007, Juan Sandro wrote:
You could read the archives from a week or 2 ago under the heading: Build
your own appliance
Yap... read it, thanks
I use these deices, but I unload them entirely into RAM.
Fine.. I though about that too but what about:
- if power fails?
*shrug*
On Wed, 12 Sep 2007, Euler Pereira wrote:
Hey all!
I'm newbie in the Asterisk World but old in other telephony systems like
Lucent/Avaya, Sopho, Siemens and Linux/Unix system.
I'm in doubt, as based system, should I install Fedora, Debian,
Slackware, FreeBSD our Sun Solaris? Which is
On Fri, 14 Sep 2007, Kate Kretz wrote:
Dear Sirs,
out asterisk server has multiple network cards.
I want some outgoing calls (from several extensions) to use one IP address,
and others to go through
another address.
is there a way to achive that using asterisk ?
I doubt it, but in any
On Thu, 20 Sep 2007, Wai Wu wrote:
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The
So I have an application where the users want to divert incoming calls on
one analogue line out to another analogue line - both lines are supplied
by BT and theres a TDM400 in the box.
Call comes in, system Dial's the forwarding number and bridges the calls.
Works fine.
Until one (or both)
On Tue, 25 Sep 2007, Ben Schorr wrote:
I have a client using the Grandstream phones (not sure which model but
it looks fairly low-end) and they're lukewarm on them. The display
doesn't tilt up for easy viewing and the sound quality on the speaker
phone leaves something to be desired
On Thu, 27 Sep 2007, Anthony Messina wrote:
On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
Hi,
I have some problems and doubts connecting two asterisk servers.
I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with
On Wed, 10 Oct 2007, Raúl Gómez C. wrote:
Hi list,
I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year
2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache),
768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb
NIC for server.
.
Gordon
Thanks Gordon!
On 10/11/07, Gordon Henderson [EMAIL PROTECTED] wrote:
Yes - You should be fine. (Based on my own use of 1GHz Via boards)
However, it's OLD.
6 years old now. What's going to fail first? The drives? PSU? Fans? Are
you going to put your company's phone system which
On Fri, 12 Oct 2007, Tilghman Lesher wrote:
On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote:
Atis Lezdins wrote:
I have 8-core system that has web interface + sql + java + some other
stuff running, and at 30 simultenous calls i get loadavg maximum of 3.
I wouldn't be too happy
On Sun, 14 Oct 2007, YT Lim wrote:
I don't seem to be able to find the necessary hardware
specs for an Asterisk server.
Look more. There are 100's of pages on it. Start at
http://www.voip-info.org/wiki/
What I have in mind is a
dedicated server to serve 50 or so people. All users
will
On Tue, 16 Oct 2007, Lees, James (UK) wrote:
I am slowly getting up to speed with asterisk. This is a very basic
problem but I would appreciate any help.
I am using a small network of clients and an asterisk server. Each
client has a headset to communicate. Is there a simple way of playing
On Fri, 26 Oct 2007, Zoa wrote:
I would stay with DECT, the battery in WIFI devices only lasts a couple
of hours. (Unless you want to take the phone with you and use it on
public hotspots etc)
The battery in my UT Starcom F1000G lasts several days, as does the one in
my Nokia E90.
However,
On Sun, 28 Oct 2007, arkda wrote:
I've been looking around for an example of a method of reading back a caller
ID value, but I haven't found anything that doesn't use Festival. I'd rather
not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly
On Mon, 29 Oct 2007, Abdul wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to
keep option if 5060 is blocked we can ask our customers to use another
port.
Really?
What country?? What ISP?
This
On Sat, 1 Mar 2008, randulo wrote:
On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote:
For your own sanity's sake, steer as far away from Grandstream as possible.
The firmware is appalling and isn't improving a great deal. They make great
steps in one area while another
On Sun, 2 Mar 2008, Mike wrote:
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
Telling people what country you're in will really help here.
If you're in the UK, I'll give you a free
I've been asked to provide a system for 200 extensions, most of which will
be existing analogue POTS handsets, not IP handsets. I've not really had
any experience with large channel banks in the past (since most of our
deployments are strictly IP-only to the desk), so I'm at a loss as to
On Sat, 8 Mar 2008, Grygoriy Dobrovolskyy wrote:
I had same problem in france, not much choice, i have ordered from germany
http://www.voipango.de
This is not an advertisement i am not working for them.
Are you sure about PRI ? in europe it's bri as i heard, PRI is in usa??
We have both
Heres a weird one...
Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks
up the number in the astdb and puts an name to it. No real magic there,
and it works well.
Same macro also has parameter passed in to put a prefix on the name - this
is set in the DDI handling and
On Thu, 13 Mar 2008, Doug Lytle wrote:
Gordon Henderson wrote:
Then no amount of Set(CALLERID(name)=somethin) will work. Even if I
explicitly do a Set before the dial, it seems to get ignored.
Trying doing that while using:
SetCallerPres(allowed)
Within your dial plan
Well there you go
On Fri, 14 Mar 2008, Eric Rees wrote:
I am having a strange issue with setting the incoming caller id on the
latest version of TrixBoxCE. Right now I have it setup with a
cross-over T1 cable to our production Asterisk (1.0.9) box and from the
Trixbox we can send and receive calls just
On Tue, 18 Mar 2008, Steve Totaro wrote:
Why not try a different OS such as CentOS for now? That would be my next
step.
I wouldn't suggest chasing distros is the way to solve issues, especially
if you're happy with the hardware.
Personally, I'd go back to Debian, but stick to stable (Etch)
On Tue, 18 Mar 2008, Paul Goodyear wrote:
Hi,
I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there
are 3 BT lines connected directly to these ports.
One of the lines has BT FeatureLine Compact and this is the line I am
having problems with, the other 2 lines are working
On Sun, 25 Mar 2007, Gordon Henderson wrote:
and change the Optional Rule to:
3,3,7,1,0;10,4,7,2,0;60
someone correct me if I've goofed!
Well, it's been a year since I wrote this, and of-course I goofed and
no-one corrected me ;-)
Bother.
So I've looked again.
In the UK, We Spring
On Wed, 19 Mar 2008, David Quinton wrote:
On Tue, 18 Mar 2008 14:50:52 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
BT nearly always try to sell featureline on business lines these days.
Would sir like a 3 of 5 year feature line contract?
Fair point, Gordon.
But in their defence
On Wed, 19 Mar 2008, Paul Goodyear wrote:
Yeah, I came accross that post too I think :) but as above, I already
tried moving the cables round, but no change.
Does anyone have a simular setup and can confirm theirs is fine?
4 Lines
1 x ADSL and Fax
3 x Voice
1 x Voice fails to answer
On Wed, 19 Mar 2008, Norman Franke wrote:
As for why a company would purchase hard phones, several reasons. First, we
are replacing many hard phones with computers. We have a custom application
and have been moving folks main numbers to use the computer. We can make it
ring externally and
On Wed, 19 Mar 2008, Senad Jordanovic wrote:
And yes, running the whole thing from standard PC based desktop will
eventually cause issues hence an solid state appliance is a way to go :)
My gripe is that I think people try to put too much into a system, don't
have a server build and
On Wed, 19 Mar 2008, Tzafrir Cohen wrote:
On Wed, Mar 19, 2008 at 06:54:46PM +, Gordon Henderson wrote:
On Wed, 19 Mar 2008, Senad Jordanovic wrote:
And yes, running the whole thing from standard PC based desktop will
eventually cause issues hence an solid state appliance is a way to go
On Wed, 19 Mar 2008, Bill Andersen wrote:
Thank you to everyone that replied to my post. I started to
reply to most of them, but it is getting a little out of hand.
Again, thank you. It actually makes me think the problem is not
so much with Asterisk as it is with implementation. (My
On Thu, 20 Mar 2008, Norman Franke wrote:
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote:
Sure some others on here may disagree, but I am also over on the trixbox
forums, and have often seen talk about the 2.6.9 kernel having interrupt
issues, and such that cause asterisk issues. One
On Wed, 19 Mar 2008, Norman Franke wrote:
On Mar 19, 2008, at 2:48 PM, [EMAIL PROTECTED] wrote:
My mobile does not sound terrible, does not have echo, does not fade in or
out, and the last time I used it to call the emergency services, I got
through straight away. I've not had a dropped call
On Fri, 21 Mar 2008, John Faubion wrote:
are plenty of phones on the market which do SIP now - most
modern Nokias do. I use an E90 Communicator, but the E95 is
popular too, so I'm experimenting with using my mobile as my
one phone, via Wi-Fi/SIP when I'm in the home/office and
Out of
On Sat, 22 Mar 2008, Alan Lord wrote:
Hi all,
I am close to purchasing some new DECT phones for our home office here
in the UK.
We use Asterisk and I am sorely tempted by the Siemens C475IP or the
soon-to-become-available-in-the-uk S685IP.
Both systems have great feature sets and,
On Sat, 22 Mar 2008, Alan Lord wrote:
Gordon Henderson wrote:
snip /
Anyone got any skeletons on them?
I've deployed a number of Siemens C460IP's.
They're really good and coverage is excellent, but for one thing: They
base stations lose registration with the asterisk box after some time
On Sun, 23 Mar 2008, Robert Lister wrote:
On Sat, Mar 22, 2008 at 09:39:47PM -, Chris Bagnall wrote:
Canÿÿt comment on the C460, but the S450 definitely doesn't have these issues:
- No SIP call transfer feature (that I can find)
Hit ext call during a call, create a new call, then you
On Sun, 23 Mar 2008, Pete Kay wrote:
Hi
I got my hylafax running to receive and send fax. Since I only have one
number, I want to know if there is anyway to detect if a call signal is fax
then redirect it to fax-to-email, otherwise route the call to my analog
phone? Is it something that
On Sun, 23 Mar 2008, mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to obtain
the Caller ID if the calls are from the phone line.
exten = s,1,Answer()
exten = s, n,
On Mon, 24 Mar 2008, mark morreny wrote:
What I need to do is to try to route called based on the dialed number as I
have multiple DIDs on my line. Is this something that can be done? Is this
something to do with the hardware that I am using? If so, what kind of
hardware do I need to
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to
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