John Novack wrote:
And CERTAIN people can post commercial stuff, if they are in favor,
while others get chastised immediately for something that may or may
not be commercial.
While we're at it, how about the seemingly endless postings of my (
fill in the blank) provider is not responding,
Hi, all
Suddenly I started to have bad sound quality. Happens with all
providers as well as with softphones connected to my * server on the
Internet.
It was OK on 1.1.17, then I migrated to 1.2.5 and it was OK for some
time. Nothing has changed in my setup, but voice quality degraged
greatly.
On Fri, Apr 14, 2006 at 05:47:32PM +1000, Rudolf Ladyzhenskii wrote:
Hi, all
Suddenly I started to have bad sound quality. Happens with all
providers as well as with softphones connected to my * server on the
Internet.
It was OK on 1.1.17, then I migrated to 1.2.5 and it was OK for some
I think you are right. 1.0.7
I connect via VoIP providers -- via Internet only. No direct PSTN
connection. (Well I do have TDM400, but did not have time to set ot up
yet).
I use Polycom SP300 phones
I even have problems when talking to people with softphones registered
on my * server.
Somehoe, I
I have upgrade Cisco 7970 on SIP using configuration file that was sent on the
list. Now, phone tries to register on Asterisk but always fails. I have sniffed
for packets with ethereal, and this is what I have found out.
First, 7970 tries to register with *.
* reply's that it's trying
* reply's
You can try:
http://www.paskambink.lt/mcc
Regards/Pagarbiai,
Mindaugas Kezys
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Wednesday, April 12, 2006 3:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] billing with
Hello,
You can try: http://www.paskambink.lt/mcc
Regards/Pagarbiai,
Mindaugas Kezys
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Tuesday, April 11, 2006 9:55 AM
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Hi everybody,
I bought a foxboard an embedded device with an
axis processor, I'd like to cross-compile Asterisk for the foxboard on
my Debian box. I use a software development kit from Axis and I have a
little tutorial from the board manufacturer on how to cross compile
a little hello world
Hi list,
I am in the process of setting up Asterisk for a new office and since
this is going to be my first real installation I'd appreciate some
advice on the hardware from the real world. We will have 8 channels
(still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely
go for a
Well and that is where my love hate relationship with Sangoma is.
While their technical support seems to be very knowledgeable it
takes forever to get ahold of them! Additionally I've found 2 bugs
now in their drivers At first setup wouldn't install... it was a
bug with their setup
At least Digium lets you wait in a queue and picks up the phone when
you call for support.. with Sangoma the only way to get ahold of
someone is to:
DIAL: 1-800-388-2475... choose option 2... get message no one is
available Press * to return to main menu. Dial extension 119. get
message no
Interesting... I don't have those problems with the A200D. In fact,
haven't found any problems at all in about two months of production use.
I have noticed that mixing a TDM400 with a A200D in the same box has an
issue with sangoma code disabling echotraining=800 (needed for the TDM).
But
Have you tried putting a Hangup in your extensions.conf?On 4/13/06, Min Hwan Chang [EMAIL PROTECTED]
wrote:Details:Asterisk 1.0.9 Zaptel 1.0
Dell P3 1ghz with X100P Clone Location: IndiaThis is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel
My main concerns would be, can you have multiple cards like this on a
system, for example, I now have a te110p and 2 tdm04b and Im getting
irqmisses on the te110p (according to zttool and zttest) which makes fax
receiving on the te110p almost impossible.. Plus, voice is getting frame
slips.
I was
I must agree with you. I too buy Digium cards because I want to support the
development of asterisk. Asterisk is a great product but digum cards are a
pain, they say they don't support faxing but a lot of people that are
implementing asterisk demand or need faxin as a day to day service on
their
Problem is, how to make sure you system WILL have 100% on zttest before
buying the cards.. You need to have stability, compatibility and certainty
that what you buy is going to work :(
Anybody had similar problems or success stories with sangoma cards?
|-Original Message-
|From: [EMAIL
Aaron, have you tried using 1 te110p and 2 tdm04b on the same server?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Aaron Daniel
|Sent: Thursday, April 13, 2006 7:19 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re:
Anton Krall wrote:
I must agree with you. I too buy Digium cards because I want to support the
development of asterisk. Asterisk is a great product but digum cards are a
pain, they say they don't support faxing but a lot of people that are
implementing asterisk demand or need faxin as a day to
Hi,
I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my
intention is to use a TDM2400P echo cancel module). It TDM2400p working
good with asterisk 1.2.1? Or I need to install a new asterisk version?
TIA
Giorgio Incantalupo
___
Simone wrote:
Hi list,
I am in the process of setting up Asterisk for a new office and since
this is going to be my first real installation I'd appreciate some
advice on the hardware from the real world. We will have 8 channels
(still not sure if 4xISDN2 or ISDN30 8 channels, but I will
Giorgio Incantalupo wrote:
I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my
intention is to use a TDM2400P echo cancel module). It TDM2400p working
good with asterisk 1.2.1? Or I need to install a new asterisk version?
There is no reason not to upgrade to the latest Asterisk
I'd have to guess that combination of cards with almost any mobo would
be considered an overloaded system. If you replaced the two TDM04b cards
with an A200D or TDM2400 card, most of those irqmisses (etc) would
probably go away; but that's a somewhat educated guess on my part.
Factually, the
On 14 Apr 2006, at 11:29, Simone wrote:
Hi list,
I am in the process of setting up Asterisk for a new office and
since this is going to be my first real installation I'd
appreciate some advice on the hardware from the real world. We will
have 8 channels (still not sure if 4xISDN2 or
I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment. When it is implemented, that card should work. The TDM400
card will not
I'm speculating here... but why does Sangoma need to patch zaptel
source?!? Can't they just use the same interfacing that the digium
cards do?
On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:
Interesting... I don't have those problems with the A200D. In fact,
haven't found any problems at all
Title: [Asterisk-Users] Will VoIP ITSP's be Next?
Comsumer acceptance of VoIP has caused the TelCo's to
rethink their profitablility, but that so no secrect to those on this
list.
It would appear that the treat is real enough for
others, such as Googleto start taking preventive
measures
Anton Krall wrote:
Problem is, how to make sure you system WILL have 100% on zttest before
buying the cards.. You need to have stability, compatibility and certainty
that what you buy is going to work :(
Anybody had similar problems or success stories with sangoma cards?
Running zttest on my
Hi Kevin,
I know upgrading is better, sorry, maybe my question was malformed...the
exact question is which is the minimum asterisk version supporting
TDM2400P?
(I have 10 pbx and I want to change 3 TDM400P with one TDM2400P on
every pbx without reinstalling a new asterisk version on every
Hello!
What do I need to use the asterisk on my notebook with a bluetooth
headset?
Is there anywhere a good howto?
Thanks!
Andi
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
Hi,
is it possible to remove the no timeout combo box in flash operator panel?
How can I reduce the flash area? I set small buttons and half of the
area is white and I want to resize it.
TIA
Giorgio Incantalupo
___
--Bandwidth and Colocation
I think that the BT channel in Asteisk is to support a phone connection
via Bluetooth, (ie asteisk mimics your Headset, and a bit more).
If you have a Bluetooth chipset in your laptop or add one via USB. Any
softphone that can you the audio to/form your headset should work, I do
not know if ALSA
On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:
I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment. When it is
Giorgio Incantalupo wrote:
Hi Kevin,
I know upgrading is better, sorry, maybe my question was malformed...the
exact question is which is the minimum asterisk version supporting
TDM2400P?
(I have 10 pbx and I want to change 3 TDM400P with one TDM2400P on
every pbx without reinstalling a new
Rusty Dekema wrote:
If this works, I don't see why a fax transmission wouldn't work. Is it
because the fax protocol doesn't have error correction? Is that even
true?
FAX transmission is massively more complex than modem transmission. At
higher speeds, it involves 3 or 4 different 'carrier'
My bluetooth headset works with the snd-bt-sco module. I also compiled
chan_bluetooth into my asterisk but I can't find any howto for using a
bt headset with asterisk.
Does this just work with a smartphone as gateway? Is it not possible
to connect the bt headset directly to my notebook?
Andi
On
Rusty Dekema wrote:
On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:
I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment.
Peter J Dean wrote:
We do it slightly different, rather than multiple macros, we do it
within a single macro.
Peter,
I have to this some questions:
1. I have not seen n(tryiax01) construction before. Can you explain
it, please and how you give this to the macro?
I know only exten =
Hi there,
If there is anybody on-list looking for VoIP related work in India, please
contact me off=list with your details.
Positions are of a full-time nature.
Regards,
Sahil Gupta
VoiceValley
___
--Bandwidth and Colocation provided by
I want to upgrade * this weekend.
What can I prepare? What will I have to change in the settings? Where
can I read about it?
I use now:
*CLI show version
Asterisk SVN-trunk-r8447M built by root @ on a x86_64 running Linux
on 2006-01-25 15:33:01 UTC
bye
Ronald
Small update, I've been able to sort of work around the problem by making the
AgentcallbackLogin() direct to a context that in turn does another dial over a
local channel with the /n that gets around part of the problem. Still kinda
nasty seeing 5 channels around for 1 call...
--johann
Hi everyone,
On the Polycom 601 phones we are using, the forward feature works very
nicely for agents that are out on trips. I was wondering if there is a
way to test to see if they have the forward option enabled.
When it is enabled the call comes in and gets -- Got SIP response 302
Moved
Hi all,
I urgently need a solution in a part of a project.
I appreciate all types of help.
The thing I absolutely need is. To play a background
music in call.
If I have the opportunity to stop it via entering a
dtmf combination is would be very very nice also.
Does anybody know some
Ronald Wiplinger wrote:
I want to upgrade * this weekend.
What can I prepare? What will I have to change in the settings? Where
can I read about it?
I use now:
*CLI show version
Asterisk SVN-trunk-r8447M built by root @ on a x86_64 running Linux
on 2006-01-25 15:33:01 UTC
As Kevin as
Has anyone been able to send a fax through a Unicall channel? I am
unable to send or receive faxes using either rxfax or a fax machine connected
to an ATA. Can someone point me in the right direction?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de
I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going on.
I can run
Hi List,
Not sure if this is the place for this so here goes ...
We have a number of Polycom 501's connected to our * box and they work
great. Some of our users have added a few entries into the directory on
the phone. The problem is on those particular phones they now sometimes
get resource
On 4/14/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
We have a number of Polycom 501's connected to our * box and they work
great. Some of our users have added a few entries into the directory on
the phone. The problem is on those particular phones they now sometimes
get resource full on
My customers are reporting that the contact directory can only hold
about 45+ entries.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by
On Fri, 2006-04-14 at 13:08 -0700, Mindaugas Kezys wrote:
Hello,
You can try: http://www.paskambink.lt/mcc
Or can try http://www.asterisk2billing.org/ it supports postgresql
Regards/Pagarbiai,
Mindaugas Kezys
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi Kevin,
I'm sorry..I was wrong again...when I wrote Asterisk I meant Zaptel (I
always use Asterisk x.y.y + Zaptel x.y.z + Libpri x.y.z, same version
for all!)
By the way your answer satisfy me. I already use 1.2.x zaptel driver. ::)
Thanks again !
Giorgio Incantalupo
Kevin P. Fleming
On 04/14/06 20:05 Matt said the following:
When it patched the
zaptel source... if I have usecallerid=yes on then it crashes... if I
turn usecallerid=no then it is fine.
we've tested the sangoma A101, A102 and A104 cards with usercallerid=yes,
and it hasn't crashed. this is on FreeBSD
Giorgio Incantalupo wrote:
I'm sorry..I was wrong again...when I wrote Asterisk I meant Zaptel (I
always use Asterisk x.y.y + Zaptel x.y.z + Libpri x.y.z, same version
for all!)
FYI... those version numbers are no longer kept in sync. The Zaptel and
libpri version numbers are incremented only
Michael Strelnikov wrote:
I just never used one. Is BIND good enough?
dnsmasqd is quick and easy. All the joys of DNS caching without the
pain of configuring a full-blown bind. Unless, of course, you do this
sort of thing every day ;-)
___
Jay Milk wrote:
Michael Strelnikov wrote:
I just never used one. Is BIND good enough?
dnsmasqd is quick and easy. All the joys of DNS caching without the
pain of configuring a full-blown bind. Unless, of course, you do this
sort of thing every day ;-)
Ryan Amos wrote:
I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going
Kevin Smith wrote:
Hi everyone,
On the Polycom 601 phones we are using, the forward feature works very
nicely for agents that are out on trips. I was wondering if there is a
way to test to see if they have the forward option enabled.
When it is enabled the call comes in and gets -- Got SIP
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem with
spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
patch from the same directory.
When starting asterisk I always get
did you try to recompile the plugin?On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfaxdoesnt work any more.I've installed spandsp-0.0.2pre25 (the same problem withspandsp-0.0.3pre6.tgz
) app_rxfax.c, app_txfax.c and made
Rob Terhaar wrote:
did you try to recompile the plugin?
yes, of course...
On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem with
I too had a server room fry and need to replace h/w.
So what specific Dell servers did/do you deploy?
Where is the link w/Digium/s Dell caveats?
I'm using the Digium TDM400 card w/*
Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT)
From: Aaron Daniel [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
after a few hours of debugging it works now...
I got some version mixes of spandsp on my system...
sorry for the spam
tom
Thomas Artner wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same
Wai Wu wrote:
I did not install soxmix in my linux box. If you having issues with
mixmonitor, you can put both legs of the call into a conference and
record the conference
Wai,
How are you mixing the leg files? Do you run a process that moves them
to a remote box with soxmix installed?
We're using 1850's for our asterisk system right now (well, all but two,
but they'll be upgraded soon). 1U boxes that work like a dream, I think
the biggest problem I've had with them is mpg123 wouldn't compile since
they're 64bit, and that was a simple fix.
This is what Digium has to say
Hi,
Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
___
The files were never mixed until they are actually listened to (They
have 3 people, 10 hours a day listening to calling recordings), and it
is done on a separate * box. As for the drive, all I know is an separate
external unit out of the main * box. (I didn't setup the linux server
myself).
Hi!
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At this point the
Hi,
Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
Why don't you download the package from the
Wai Wu wrote:
Hi,
Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the
call
.) the caller get lost at this point !!
At this point the
Michael Collins wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the individual
hard phone.
This of course is if using SIP which we do not know yet...
On Apr 14, 2006, at 1:43 PM, John Novack wrote:
Michael Collins wrote:
A few
So, what version of spandsp are using afterall?
[]'s
MM
-Original Message-
From: Thomas Artner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Fri, 14 Apr 2006 19:50:47 +0200
Delivered: Fri, 14 Apr
Jerry Jones wrote:
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the individual
hard phone.
Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There SHOULD
Jerry Jones wrote:
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the individual
hard phone.
Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There
On Fri, 2006-04-14 at 08:19 -0500, Rich Adamson wrote:
I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment. When it is
Keep in mind that with a SIP phone you are not communicating directly
with asterisk but with the phone which acts on your behalf with
asterisk. On traditional systems if you performed a hook flash to
transfer, you were definately signalling directly to the PBX. Now
when you push a button,
on my system, if i do a blind-xfer, it rings the destination's phone and finally flips to voicemail. Sometimes, If the destination/recipient is an exec or otherwise important, our attendant does a normal xfer to see if they're at their desk, if the destination doesn't respond, then the attendant
previously, i've found directions on how to get a bluetooth headset to act like a sound card in windows. Can't remember the URL though...On 4/14/06, Andreas Nitsche
[EMAIL PROTECTED] wrote:
My bluetooth headset works with the snd-bt-sco module. I also compiledchan_bluetooth into my asterisk but I
Shaun schrieb:
Anybody know the proceedure to factory reset the a 7960 phone running 6.3
SIP software? I've tried holding # when booting the phone and nothing, i
can do that on my 8.2 phone but this phone i just got with 6.3 isnt working.
Also **# doesnt work either..
Hi Schaun,
i have
There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom phones
not being able to do a blind xfer using the feature key.
We have to use the asterisk # blind xfrer functionality for blind
transfers
The phones will drop
Hello:I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic.Is anyone using just asterisk for production purpose. Meaning serving a high number of callers.
Is it mandatory to use SER
On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:
Some people have problems, some people don't. There is no way you can be
prepared for every situation out there. We try our best.
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using
Jeff Gustafson wrote:
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems. Does
this newer bus standard help the situation with faxing?
No. PCI-X is just a wider/higher-speed version of PCI, not a new
Well, the TE410P and TE411P work in the PCI-X slots since it's backwards
compatible. So I guess in effect, the Digium's cards already do support
it :)
Aaron
On Fri, 14 Apr 2006, Jeff Gustafson wrote:
On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:
Some people have problems, some
On Fri, 2006-04-14 at 15:10 -0500, Aaron Daniel wrote:
Well, the TE410P and TE411P work in the PCI-X slots since it's backwards
compatible. So I guess in effect, the Digium's cards already do support
it :)
My fault. I meant to say PCI-e, which is a newer bus that Dell is
shipping
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between some SIP
phones and asterisk, I have personal experience with Polycom phones not being
able to do a blind xfer using the feature key.
Is that a Polycom or Asterisk defect?
We have to use the asterisk
Michael Collins wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the
call
.) the caller get lost at this point !!
Melcon Moraes wrote:
So, what version of spandsp are using afterall?
i am using spandsp-0.0.2pre25 now.
In the 0.0.3 package, there is no app_rxfax.c and no app_txfax.c. No
idea why thats missing there.
tom
[]'s
MM
-Original Message-
From: Thomas Artner [EMAIL PROTECTED]
On Fri, 2006-04-14 at 15:10 -0500, Kevin P. Fleming wrote:
Jeff Gustafson wrote:
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems. Does
this newer bus standard help the situation with faxing?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeff
Gustafson
Sent: Friday, April 14, 2006 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
sodisappointing !)
On
Jeff Gustafson wrote:
My fault. I meant to say PCI-e, which is a newer bus that Dell is
shipping on their server class machines.
Right. That is not supported by any Digium products yet, but it still
won't help the FAXing issue, since the issue is _not_ PCI bus bandwidth.
In fact, the
Not sure, but the fact that the # xfer in asterisk releases the call
without the ability to do an attended transfer is an asterisk issue,
maybe not a defect, but a design issue inconsistent with typical PBX
behavior.
To be typical it would act like this;
Press pound to get secondary dial tone
Jeff Gustafson wrote:
Is there any reason an easier implementation of the same, basic, idea
could be created for the Asterisk generation? According to a quick
search of H.100 it's just a TDM bus. It handles 2,048 full duplex
calls. Would a lightweight version that only supports 512
I sent the following message a few days ago, but never received a
reply, so I thought I'd ask again..
Can anyone tell me how me to get asterisk to dial out a phone number using BTP
when a bluetooth device is not detected? I can get BTP to dial to a
SIP phone, but I can't get it to dial through a
On Fri, 2006-04-14 at 15:35 -0500, Kevin P. Fleming wrote:
Jeff Gustafson wrote:
My fault. I meant to say PCI-e, which is a newer bus that Dell is
shipping on their server class machines.
Right. That is not supported by any Digium products yet, but it still
won't help the FAXing
Hi everybody,I would like to be awareabout what happened to me.Two weeks ago, on a Sunday morning a French guy called me. Ask me to fix some problems with his asterisk. After fixing his problem, he asked more and more, after 10 hours of work I ask him to pay me for the first milestone. However,
Well... did you tell him your services where not free and come to a
financial arrangement before you started?
-Original Message-From: Voce Lavoce
[mailto:[EMAIL PROTECTED]Sent: Friday, April 14, 2006 3:14
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] My
Hi Douglas,sure I gave him my hour rate and he agreed.He also promised me to pay a week ago.See youOn 4/14/06, Douglas Garstang
[EMAIL PROTECTED] wrote:
Well... did you tell him your services where not free and come to a
financial arrangement before you started?
-Original
That's the nature of consulting - you have to balance
demonstrating competency with solving the problem before being paid. We've
had many similar experiences, and we now require prepayment for 2 hrs service
before we do any work (or even talk to the client for more than a
fewminutes).
Just saw this on Cisco's software download site:
7941/61 IP Phone SIP phone load - for CCM v5.0
Has anyone used this with Asterisk yet?
Josh
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