Deepak Naidu wrote:
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D
card. To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX
setup.
It would help to know exactly what Dell Poweredge you were considering.
They do vary.
If you compile your kernel
Corporate IT Solutions - Michael Dunne wrote:
I have now within 18 months had a second TDM400P die, the first time was
random call drops, and now it will not go off hook when making a call.
To summarise, the card stopped making calls, I replaced the computer
hardware, installed new OS and new
[EMAIL PROTECTED] wrote:
Thanks for the reply. Unfortunately that didn't work. What's confusing
is that for the line without any distinctive ring that works correctly
with callerid, the only thing it does is dial the phones, so here's the
entire context:
[add-incoming]
exten =
Erick Perez wrote:
Hi,
after many issues we finally managed to make our system do outgoing
calls with perfect quality.
However I cannot detect *any* form of incoming call. when I use an
outside phone to call the E1 connected to the sangoma a102, I
instantly get a fast busy tone.
Let's see
Erick Perez wrote:
Yes I do. I even did a pri debug span 1 and when I call the asterisk
box, it sees nothing.
Hmn, well, that's telling.
Are you using the correct cable? Is the cable plugged into the correct
port on the card? The 102 is a two-port.
-Stephen-
Cheikhou DIAW wrote:
hi , i think everybody is receiving theses mails from rp.
can someone unsubscribe or do something , its really annoying
Now I know why I had 600 messages in my Asterisk folder after only three
days away.
-Stephen-
___
Patrick Buller wrote:
Nope, that's good. It means you have registered to their server no problem.
Firstly, which version of Asterisk are you using?
Version 1.2.7
That is super old. Did you install it from a package? I recommend you
upgrade now, because you will have to later, I
Dave Donovan wrote:
On 7/10/07, *Stephen Bosch* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Jason Aarons (US) wrote:
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN
Walt Reed wrote:
On Wed, Jul 11, 2007 at 09:34:51AM -0600, Stephen Bosch said:
Walt Reed wrote:
No, as I explained before with the reasons why, please don't post them
here. Send them DIRECTLY to the list admins. It is 100% off topic to
keep discussing a list administration / mail delivery
Hi, Jakub:
Jakub Głazik wrote:
Asterisk [EMAIL PROTECTED]
Client hears pure silence when waiting for call answer. Music on hold stops
when transferer pics a number and client doesn't even hear ringing.
Is this normal behaviour? How to change this?
Sadly, this is normal behaviour.
Log
Walt Reed wrote:
No, as I explained before with the reasons why, please don't post them
here. Send them DIRECTLY to the list admins. It is 100% off topic to
keep discussing a list administration / mail delivery problem here.
List USERS can not help you.
Considering that the vast
Jason Aarons (US) wrote:
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN BRIs for voice or data since early 90s.
I don't know what it's like in your area, but here, fractional PRI is
just not cost
Patrick Pfeifer wrote:
Hello,
I was wondering if there is a way to change the From address (not just
the Return-Path) for voicemail notification emails in Asterisk.
It looks like the serveremail directive in voicemail.conf just changes
the Return-Path.
I'm looking for something
Jon Pounder wrote:
http://www.crtc.gc.ca/archive/ENG/Orders/2007/o2007-56.pdf
some discouraging directions being taken by the idiots at the crtc.
Essentially laying the groundwork to phase out bri completely in
Canada, probably fcc has similar idiots making similar decisions as we
Jeff Davis wrote:
If there was a driver available, I'm still not sure how many installs I
could sell. Verizon wants to pretend the service doesn't exist, and the
largest CLEC in my area doesn't even sell it. (I even offered to buy my
CLEC rep dinner and she wouldn't sell it to me.)
Wayne wrote:
I was wondering where 3Com were getting all the new ideas from for their
phone system ;-p
Cats out of the bag now I guess :)
The price of open source is that the commercial outfits are free to rip
off ideas without paying for them.
But hey -- competition is good, right?
Jeff Davis wrote:
Stephen Bosch wrote:
Your rep at Sangoma? Or your reseller?
That wasn't very clear. Sorry. It was Sangoma.
(I would be more verbose, but I don't want to spam the list)
I just wanted to make sure it wasn't stale information.
This is a real chicken-and-egg problem. More
Jon Pounder wrote:
most of the first level reps I have ever talked to in the last 10
years don't even know it exists, higher level people claim they don't
offer it, still higher level people know what you are talking about
when you say its tariffed and finally cave in to what you want.
Dave Donovan wrote:
On 7/5/07, *Stephen Bosch* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I would be willing to help out with a driver, but without a line and
card I am not sure how productive that would be.
As I've already said, I can get one, and it's not a big deal
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the basic signalling could
probably be tested with it.
is exploring some sort of back to back
Jeff Davis wrote:
I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the
line simulator and a phone would work. Then get a BRI line when there's
a driver that looks like it works.
You'd think it would -- otherwise the line simulator is somewhat
pointless, isn't it?
I saw
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who wants it
for play?
Well, whoever ends up with the simulator should get
Mark Phillips wrote:
Damn!!! Beat me to it ;-}
As an Englishman now living in New Jersey (strangely nowhere near an
exit) I have to say that the local idiom and accent leaves a significant
amount to be desired.
Terms like New Joisey, Shuwa ,wadder, badderies,
congradulations etc make me
Andrew Kohlsmith wrote:
On Tuesday 03 July 2007 9:47 pm, Joe acquisto wrote:
We get to do that, because, back in the late 1700's . . . we won.
Hey man, I'm Canadian... We've got our own set of funny accents, and don't
get
us started on the Quebecois. Not even the Parisians can understand
Jaswinder Singh wrote:
Think about voicesense which will sense what you are talking and pop in
a *relevant* voice ad to spice up conversation :P .
If this happens I am going back to tin cans and string.
-Stephen-
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Doug Lytle wrote:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
list seems fine!
Mine arrive instantly.
-Stephen-
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Dave Donovan wrote:
Sorry I'm a little late to the thread but this question has puzzled me
as well. My key thing for me is hardware.
On 6/27/07, *Joe Greco* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Thoughts? Who here has used BRI in North America? And when you
did,
Jon Pounder wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote:
Sorry I'm a little late to the thread but this question has
puzzled me as
well. My key thing for me is hardware.
In the UK but ... Cheap BRI card...
To use it in the
Jon Pounder wrote:
Quoting Dave Donovan [EMAIL PROTECTED]:
Sorry I'm a little late to the thread but this question has puzzled me as
well. My key thing for me is hardware.
I hear you loud and clear - I am in much the same situation.
from what I know about the cards (which isn't much)
Tzafrir Cohen wrote:
On Wed, Jul 04, 2007 at 12:46:35PM -0400, Jon Pounder wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote:
Sorry I'm a little late to the thread but this question has
puzzled me as
well. My key thing for me is
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
http://www.sangoma.com/datasheets/A500BRI
is that the card you mean ?
it says it supports asterisk
Yes, that's the card I mean and yes, it supports Asterisk.
The problem: I have been told -- again, this is tentative
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
http://www.sangoma.com/datasheets/A500BRI
is that the card you mean ?
it says it supports asterisk
Yes, that's the card I mean and yes, it supports Asterisk
Jeff Davis wrote:
Jon Pounder wrote:
If someone already has a customer relationship with them, ask straight
out does it work in US/Canada with the BRI available here with asterisk.
I just got off the phone with my sales rep. It appears I'm the third
person today to ask about this. (I
Chris Mason (Lists) wrote:
I love the smell of lemonade in the morning
Do lemons grow in backyards in Anguilla?
-Stephen-
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Eric ManxPower Wieling wrote:
John Faubion wrote:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
I've seen several people mention it taking a few days to send messages. I've
usually seen mine in a
Ade Vickers wrote:
Hi Richard,
Thanks for those replies - I'll give them a shot shortly.
That's not really what I meant by configuration -- you can choose the
MOH source for Asterisk. It's only the native player that restarts the
music file every time someone is put on hold.
We're still
J. Oquendo wrote:
Karl J. Vesterling wrote:
Actually, I *NEED* to change the caller ID. Here's why...
CID internal and external are two different things.
I think Karl was referring to external caller ID.
If PSTN gateway providers lock the callerid to my DID and I have no
way to change
Eric ManxPower Wieling wrote:
Maybe Digium can start using 1.4.x on THEIR production boxes like the
Digium corporate PBX and IAXTel.
Not doing so is like the city water department saying that the water is
safe to drink, but every employee of the city water department drinking
bottled
Russell Bryant wrote:
Lacy Moore - Aspendora wrote:
On 6/29/07, Ade Vickers [EMAIL PROTECTED] wrote:
What I'd like to do is have the music streaming constantly, so the on hold
caller always gets music at the current position; even if that's in the
middle or near the end of a file.
Many of
Danny Brown wrote:
I have been trying for a very long time to get asterisk to detect and
utilize dtmf tones from my sip clients within my dial scripts. I have
set automon=#9 in my features.conf, I have Dial(,gWw) in my dial
scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script
Asif Raza wrote:
hi,
i am using Asterisk 1.4. and unable to get Voice Mail below is my config
extensions.conf
exten = 50,1,NoOp(Failover)
exten = 50,2,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = 50,3,Dial(SIP/101,18)
exten = 50,4,Goto(ss-${DIALSTATUS},1)
exten =
Al Bochter wrote:
Well the gun owner will go to jail!
Take a look at your local news.
If you own a gun, it's your responsibility to keep it secure. I don't
know of an OECD juridiction where that's not the case.
-Stephen-
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Jeremy Mann wrote:
you would think the telcos would be more interested in selling this
to small/medium businesses that are not ready for a voice pri but
it
Since when to the telcos have the consumer's best interest in mind?
They can sell you a PRI at full loop cost with a smaller number of
Hi, folks:
I remain intrigued by the gap in BRI implementation between North
America and Europe, and I wanted to get feedback from the list members
on the matter. I'm seriously considering making the leap in our office.
In Europe, the idea that an office that does not have enough lines to
Greg Oliver wrote:
On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote:
Hi, folks:
Snip
Thoughts? Who here has used BRI in North America? And when you did, what
interface hardware did you use?
-Stephen-
I grew up on BRI when the internet first started taking off here. All
Thanks for the response, Joe.
Joe Greco wrote:
Voice BRI is scarcely advertised. In these parts, Telus does indeed
offer it. (I had to know what I was looking for, though.)
BRI is a service the telcos would like to forget about here in the US.
We ordered it at the house because we're
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they
John Novack wrote:
In this troubleshooting case, it probably is better that there is NO
dialtone, which would make the hiss easier to hear.
I am curious what the OP found
When Asterisk is stopped, does the hiss continue?
Okay -- I have tested this and yes, the hiss is still present even after
John Novack wrote:
In this troubleshooting case, it probably is better that there is NO
dialtone, which would make the hiss easier to hear.
I am curious what the OP found
When Asterisk is stopped, does the hiss continue?
That's tough to assess because the other problem I have had with it is
Hi, John:
This feedback is brilliant. Thanks. My comments follow.
John Novack wrote:
In your case, from listening to the recording, it really seems as if it
is being generated within the card. To be sure, if you haven't already,
connect to the FXS port directly from a telephone with one of
Tzafrir Cohen wrote:
To generate a FXS dialtone without Asterisk, use fxstest (make fxstest)
from the zaptel source directory.
Can I break this dial tone with DTMF?
-Stephen-
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shadowym wrote:
Every Sangoma A200 card I have ever connected to the PSTN required a rx gain
of at least 10. Yours is commented out which I believe would make it
default to 0?
The noise is present on FXS ports only and is audible the moment the
receiver is lifted.
I mention this because
Hi, Jorge:
Jorge Mendoza wrote:
Never experienced with FXS modules on a PC with Asterisk. However we
have experienced that kind of problems on legacy PBX without a good
ground. If you replace the system with a analogue set and have not
noise, then a ground current is generated in your
shadowym wrote:
All I know is that your rx gain, at 0db is probably way too low. My tx gain
is typically set to -3db.
I set it 10 as per your previous post and ended up with audio so loud
there was clipping and distortion. I don't think that's the cause.
Sangoma has expressly told me to
John Novack wrote:
Since the OP said the noise was on FXS ports, Jorge's answer isn't
relevant.
After listening to a wav file of the noise, it sounds to my old ears
like a background hiss or so called comfort noise, except for a couple
of short pops which I assume to be an open microphone.
Gavin Henry wrote:
Dear all,
We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes.
SIP Inbound
shadowym wrote:
I don't see any mention of you adjusting gains on the card/phones. Also,
what are you doing for echo cancellation? Can you post your zapata.conf
file?
I had actually tried to adjust the gains, but it actually seemed to make
the problem worse.
I turned echo cancellation off
Fuermann, Jason Bryce wrote:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html
This only works if you have a reseller account.
-Stephen-
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Andrew Latham wrote:
Tzafir
Most small/medium companies have a T1 for all their phone needs.
Internally there is a need for some analog lines.
* Fax Machine - FXS
* Security System (most ask/demand two lines) FXS
* Paging - FXO
* Dialup systems
I think he's asking why both T1 and FXS/FXO
Hi:
I have a Sangoma A200 card installed in a server with two FXO modules
and one FXS module.
Analog sets connected to the FXS module have a squeaky static -- it's
like static mixed with the sound of someone vigorously cleaning a window
a few doors down.
In other words, it's not a classic
Stephen Bosch wrote:
Hi:
I have a Sangoma A200 card installed in a server with two FXO modules
and one FXS module.
Analog sets connected to the FXS module have a squeaky static -- it's
like static mixed with the sound of someone vigorously cleaning a window
a few doors down.
In other
Stephen Bosch wrote:
Stephen Bosch wrote:
Hi:
I have a Sangoma A200 card installed in a server with two FXO modules
and one FXS module.
Analog sets connected to the FXS module have a squeaky static -- it's
like static mixed with the sound of someone vigorously cleaning a window
a few
Rob Schall wrote:
Are you able to access the phone via a web browser? And did asterisk
register the phone? If both are true and you set the always reboot flag
to 1, then rebooted the phone by hand, there shouldn't be anything
standing in the way.
It seems that it will only reboot if certain
Hi:
[EMAIL PROTECTED] wrote:
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
This will not work with this channel driver. Explanation follows.
My
Ricardo Martins wrote:
Thinking this way, I invite those who think about the open source
communities just as a zero price and its mailing lists as a space to
wait passively for answers, to rethink its own ideas. Before asking for
something and adding trash to communities mailing lists, DO A
Gordon Henderson wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.
This is just with a normal Dial command.
Rob Schall wrote:
If all you need is a soft reboot to load config files and want to do it
remotely, there is no need to cut power to the phones. I know this works
on polycom 501 and 601s. I assume it would work on other polycoms as well.
asterisk -rx 'sip notify polycom-check-cfg
Savoy, Kevin - Williston, ND wrote:
Not necessarily. If you set the following in the sip.cfg file to 1 it
will reboot even if there are no changes. The default is zero which will
only reboot if there are changes.
voIpProt.SIP.specialEvent.checkSync.alwaysReboot=1/
Hoo-ray! Thanks for this!
Rob Schall wrote:
Correct. Once this is set to 1, then it will reboot regardless. I've
been using this effect for over a year.
Hmn -- just tried this. It doesn't seem to be working...
-Stephen-
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Jonathan Creasy wrote:
Which sounds like exactly what I described. Asterisk in Dom0...
Whether it's Xen dom0 or domU barely matters. You're still working with
a patched kernel. You're taking your chances.
Good luck!
-Stephen-
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Carlos Chavez wrote:
On Tue, 29 May 2007 20:20:13 -0500, Eric \ManxPower\ Wieling wrote
I just made another test by dialing to a Zap channel instead of a SIP
phone and the call goes through without any problem. It is just when
you try to dial to a SIP phone that you get the
Chris Earle wrote:
Well, yeah, I know it's do-able with either the Sangoma card or Digium's own
TDM2400 but I don't want to replace the TDM400p I've already got in
there
Anyone think two TDM400p's won't cause me any trouble?
I think I replied to this already, but I'll give it another
Steve Totaro wrote:
Rather hasty I think. I think whatever version 1.2.X winds up on should
be the most stable release of Asterisk, period.
Great -- so do I. But I'm not developing Asterisk either.
It was going to happen sooner or later -- at least this will encourage
more people to get into
John Novack wrote:
A common attitude in the development community.
Keep adding more bells and whistles, it's more fun and interesting.
Don't bother to fix the many existing problems. That is boring
You know what's more boring? Having two feature-frozen versions of the
same software fighting
John covici wrote:
I have an install using Rhino cards -- I sure hope they get their act
together by then.
They have no choice now, do they?
Nothing focuses the attention like a deadline.
-Stephen-
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Roberto Pereyra wrote:
Hi all !!!
I would like to install asterisk in Xen domU using TDM400 hardware.
Somebody know a howto or tutorial about that ?
Here is my tutorial:
1. Install TDM400 card.
2. Install Xen.
3. Create domU, install guest OS.
4. Install Asterisk on guest OS.
5. Spend the
JR Richardson wrote:
Hi All,
Has anyone gotten the polycoms or the linksys phones to accept oprtion
66 on the dhcp request for the address of the tftp config server?
We have the dhcp server issuing the proper IP of the tftp server, but
the phones just sit there and never try to contact
Hi:
Does anybody know of a TDM interface card for *digital Centrex* that
will work in Asterisk? We're not talking about BRI, here -- the lines
have Nortel digital sets on them, and we want to run them into an
Asterisk PBX.
Centrex is more widely used in NAm.
-Stephen-
Hi:
Can anyone recommend a good ISDN BRI interface card for Asterisk? I know
there are a few out there.
-Stephen-
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Andre Courchesne - Consultant wrote:
Hi,
Anyone has details or information on how to use the SMS command to
send SMS to Fido, Bell Mobility and Rogers Wireless in Canada?
Bad news on that.
SMS() requires a PSTN port or device in order to work. None of Rogers,
Fido, Bell or Telus provide
Asterisk wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
peak times, so this shouldn't cause echoes either, right? Hmmm, so
handset could be an issue, but did anyone ever experience any handset
Hi, folks:
Is there any reason why MusicOnHold() would die after 60 seconds? That
looks suspiciously like a default timeout. How can I make it indefinite?
-Stephen-
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Stephen Bosch wrote:
Hi, folks:
Is there any reason why MusicOnHold() would die after 60 seconds? That
looks suspiciously like a default timeout. How can I make it indefinite?
Moral of the story -- don't work at 4 am.
The call terminates after 60 seconds because I never answered
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting Explicit MWI Subscription to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the mailbox= variable in sip.conf? I made that mistake
yesterday and
Eric ManxPower Wieling wrote:
What are the advantages of 9.x over the 8.x that I currently use?
I was about to ask the same question. What if my 8.x EC works just fine?
(Why expose yourself to the possibility that even the patched version
fails?)
-Stephen-
Brian Capouch wrote:
Stephen Bosch wrote:
If you get dumbfounded responses ask to speak to someone in the
programming group (unless they are a tiny little phone company, they
will have one). If you open a ticket, it usually means they will
escalate the problem, even if the agent you
Brian Capouch wrote:
Stephen Bosch wrote:
And what do you do when they say:
We have a modern, relatively-new switch for which that sort of feature
change is a trivial click on a GUI checkbox. However, we do not have
any tariffed requirement to provide disconnect supervision. So we won't
Olivier wrote:
2007/5/16, Stephen Bosch [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
As you mentioned, try implementing it only on
certain channels, if you can.
-Stephen-
Hi,
Unfortunately, I understood you couldn't allocate HPEC on per channel
basis.
But anyway, I
Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]:
Hi all.
We have Snom phones which do have a defined key in order to drop incoming
call WITHOUT answering.
Pressing that key, a SIP/2.0 486 Busy Here message is sent back.
We have other
Klaverstyn, David C wrote:
The Telco in Canada is been real painful. I was wondering if anyone has
installed a Digium TE1X0P card in Canada and if their Telco was so
difficult.
Who is the telco? Where?
The Telco will not provide us a service until they see a FCC or DOC
number for the
François Delawarde wrote:
I don't really know of other virtualization technology other than Xen,
and I thank you for guiding me through this, but I have a few doubts
related to the choice of a virtualization technology in a host with
Asterisk:
- Isn't the fact that KVM is now included in
François Delawarde wrote:
And I thank you for that (the helping part), you've found the deep cause
of all my zaptel problems (Xen), so please don't leave me alone! ;-)
To be a bit more constructive, I'd like to ask you or anyone that dared
to try using Asterisk on a non-dedicated hardware,
Chris Mason (Lists) wrote:
The only thing I'd probably lose is the ability to do faxes! So I am
going
to investigate that further first!
Havn't doen that in years - an online fax service sends me my faxes by
email and I sent out faxes through them, not that I ever do that.
The
Chris Mason (Lists) wrote:
Stephen Bosch wrote:
The fax-to-e-mail services charge as much as the telco does for a
business line, sometimes more (at least, the ones I can deal with in
this area). Better to set-up hylafax, IMHO.
http://www.maxemail.com/fax/fax-lite.html
$24/annum.
I
J. David Bavousett wrote:
Problem A: Dialing in. If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context. So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute
François Delawarde wrote:
aaah...
I'm running asterisk in a Xen kernel, but not on a virtual machine
(DomU), only on Dom0, so it's supposed to be running on the physical
server (no PCI frontend device, ...). I had seen possible problems with
older versions of Xen, but only with ztdummy
Olivier wrote:
2007/5/15, George Pajari [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
If you have the clipping issue, make sure you get HPEC version 8.2
from Digium.
Note, however, that we have observed stability issues with HPEC 8.2 (two
kernel panics in
Alex Balashov wrote:
On Wed, 16 May 2007, Stephen Bosch said something to this effect:
Would this still be possible? (All these services have numbers in remote
area codes or have 800 numbers).
Can anybody suggest one that will take a ported number (in Canada)?
That's just something you
Olivier wrote:
Do you mean nobody has ever done this before (as I thought before asking
this question to the list) ?
So which tool KDE users are using for this ?
This KDE user is using vim :P
(What's wrong with vim?)
-Stephen-
___
--Bandwidth and
George Pajari wrote:
From c|net News:
On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office
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