Matt wrote:
I had read somewhere (but now can't find) that instead of a reboot I
can just unload the zap module (after stopping asterisk) and reload
it? Can anyone confirm this?
I do a nightly shutdown of asterisk, do a ztcfg -s, unload the modules,
and then fire it all up again.
cheers,
On Tuesday 03 January 2006 05:48, Paul Dugas wrote:
On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote:
I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk.
Does this unit require any funky dialing when placing outbound calls
from * through the phone? Do the docs
I am setting up 10 SNOM 320s for a customer, and there seems to be a problem
with call-limit and hints.
Here is my sip config for one phone:
[944]
type=friend
context=x
language=de
accountcode=x
notifyringing=yes
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
callerid=x 944
No, with Asterisk Manager you can grab Caller and Called ID.See Link, Ring eventhttp://www.voip-info.org/wiki/view/asterisk+manager+events
Cheers,Giovanni Miano2006/1/2, [EMAIL PROTECTED] [EMAIL PROTECTED]
:
umm - you usually grab it from the cdr...and it
works very nicely if you are
http://lcdsmartie.sourceforge.net/Cheers,Giovanni Miano2006/1/2, Matt Riddell
[EMAIL PROTECTED]:Yes, we do development under Linux for this.Was there some particular
support you were after?--Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily
Is it possible to auto-provision spa-3000's via dhcp like hardphones can?
-Dan
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Is there a way to have AgentCallbackLogin make an announcement before
requiring the callee to press #?
I can not find anything in the documentation or other sites (voip-
info etc). And at the moment the way i have it setup
AgentCallbackLogin calls the agent and waits till # is pressed, it
Hi
The new meetme i feature in asterisk1.2.1 for annoucing user join/leave
is good, but the initial steps to record the name and confirm seems lenghty,
the user shoudl just say the name and get into the conference, How can i
disable the confirmation of the name recorded before entering the
Hi there
our company can provide custom integration with every kind of LCD display
Andrea
Giovanni Miano wrote:
http://lcdsmartie.sourceforge.net/
Cheers,
Giovanni Miano
2006/1/2, Matt Riddell [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Yes, we do development under Linux for this.
Yes, there is a patch for this (search mantis), it's static in that it's
a single announcement that doesn't currently relate to the queue.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 03 January 2006 10:00
To:
Hi,
Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I have left any information out please let me know.
I have recently begun using asterisk on debian.
[EMAIL PROTECTED] /usr/sbin/asterisk -V
I'm trying to set caller presentation to
prohibited and I'm having slight problems doing
it.
Using a machine that has a Sangoma facing my Telco
works but when using an asterisk that talks to the
first machine using SIP it does not work.
I suspect that SetCallerPres is not transitive, ie
it's not
On Tue, Jan 03, 2006 at 10:47:59AM +, Ben Fitzgerald wrote:
Hi,
Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I have left any information out please let me know.
What do you see on
Hi,
I currently have a TE210P with 2 E1 lines, one of them goes to the
Telco which is fine and the other one goes to a Siemens HiPath 3750 PBX.
The problem is that signal that the HiPath return is to HIGH and
generates a lot of echo even when talking with a PAP2 on the same
subnet,
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
I don't have this in my main installation, which is 1.0.7.
In the case of 1.0.7 where else can I effect that change?
I also have a 1.2.1 setup, what would I have to change in the code below?
What is the general idea?
Indeed, this is 1.2.1
But do
Hi,
Asterisk doesnt support USB phones directly. You need a softphone and
then a compatible USB phone.
I have been looking for cheap USB phones which work with SJ Phone since
a while. Some of them are listed at http://sjlabs.com/sjp.html. Clarisys
and Eutectics are good but costly than what i
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
/blockquote
FYI, this is the relevant extensions_custom.conf entry on an AAH system:=
I'm not using [EMAIL PROTECTED] Thank you!
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
/blockquote
FYI, this is the relevant extensions_custom.conf entry on an AAH system:=
br
It works great on Asterisk 1.2.1
exten = 270,1,Answer
exten = 270,2,Playback(at-tone-time-exactly)
exten = 270,3,SayUnixTime(,/Europ/Zagreb,AdBY
Hello,
The idea is the following:
For the 1.2.1 installation just set the parameter
disconnect = *97
In your features.conf
For the 1.0.7 installation you either upgrade or patch the code. The patch
the code would require you a lot of knowledge of c programming. It would
consist of extracting
Joseph Rothstein wrote:
I am setting up 10 SNOM 320s for a customer, and there seems to be a problem
with call-limit and hints.
Here is my sip config for one phone:
[944]
type=friend
context=x
language=de
accountcode=x
notifyringing=yes
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
What is your firmware version? I have a few problems with the release 7.5
Miguel
Ben Fitzgerald wrote:
Hi,
Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I have left any information out
i have the same problem and also have submitted it as bug
http://bugs.digium.com/view.php?id=5281.
the Patch-5281-v2.txt in the mentioned bug will solve your problem.
Paradise Dove
On 1/3/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Joseph Rothstein wrote:
I am setting up 10 SNOM 320s
hi,
i would like all my calls originating from asterisk users bound for
external to route to one destination, a session border controller.
protocol used is sip.
i have edited extensions_custom.conf with:
exten = _.,1,dial(sip/[EMAIL PROTECTED])
would this be correct to send any calls from
On Tue, Jan 03, 2006 at 01:34:28PM +0200, Tzafrir Cohen wrote:
On Tue, Jan 03, 2006 at 10:47:59AM +, Ben Fitzgerald wrote:
Hi,
Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I
On Tue, Jan 03, 2006 at 09:46:32AM -0300, miguel saravia wrote:
What is your firmware version? I have a few problems with the release 7.5
It's 7.4. I have read a few comments about 7.5 so only went to 7.4:
Loadid: SW: P0S3-07-4-00 ARM: PAS3ARM1 Boot: PC030301 DSP: PS03AT45
Thanks,
Ben.
Dushyanth Harinath wrote:
Hi,
Asterisk doesnt support USB phones directly. You need a softphone and
then a compatible USB phone.
Asterisk does support the Digium S100U USB analog FXS adapter. It's
based on the TigerJet chipset found in many cheap USB phones. The S100U
looks like the stock
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
It's a common complaint.
Have you searched the archives? Look for disconnect supervision.
I have now. And things are a litle bit more clear to me. Thank you for
hint.
--
Tomislav Parcina
[EMAIL PROTECTED]
Hello,
I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three Way
Calling with a SIP or analog Phone is working perfectly.
But if I try to do Three Way Calling with my ISDN Phone I get an error
message: Facility Name requested on channel 0/2 not in use on span 1
I use bristuff
Hi,
i am trying to use the Prefix() application in my dialplan but ...it is not
there:
pbx.c:1690 pbx_extension_helper: No application 'Prefix' for extension
(test, 1233, 1)
My entry in extensions.conf is the following:
[outgoing-calls]
exten = _12xx,1,Prefix(0)
exten =
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
There is already a very good database for binary files,
called a filesystem
Is there any how-to for filesystem and Asterisk voicemail storage?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
Do you have a url for the device?
From: [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cell phone
Apologies for hitting the list with such a long mail on my first post!
Having seen the archives this seems like a list that likes debugging
output. If I have left any information out please let me know.
I have recently begun using asterisk on debian.
[EMAIL PROTECTED] /usr/sbin/asterisk
On Tue, 2006-01-03 at 14:06 +0100, Bukoka Budoka wrote:
Hi,
i am trying to use the Prefix() application in my dialplan but ...it is not
there:
pbx.c:1690 pbx_extension_helper: No application 'Prefix' for extension
(test, 1233, 1)
My entry in extensions.conf is the following:
I can not compile the h323 support for macosx 10.3?
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Simone Cittadini wrote:
What about IAX ? If I connect two asterisk servers to a common mysql
backend (only iaxusers, no sip or extensions) will it :
There is no support for sharing dynamic peer registrations between
Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime
Does anyone know whether there is some sort of time zone option? I've
emailed Aastra who didn't come back to me. I would like to set the time
zone - e.g. Britain-London, in the cfg files so I don't have to set it
on 40 phones...
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
Olle E Johansson wrote / skrev:
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is common.
We have users, what like to have more then one Phone, - Homeoffice
and Desk
On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
Does anyone know whether there is some sort of time zone option? I've
emailed Aastra who didn't come back to me. I would like to set the time
zone - e.g. Britain-London, in the cfg files so I don't have to set it
on 40 phones...
in
What about IAX ? If I connect two asterisk servers to a common mysql
backend (only iaxusers, no sip or extensions) will it :
There is no support for sharing dynamic peer registrations between
Asterisk servers via Realtime for SIP or IAX2. Sharing the Realtime
database for users and
On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote:
Does anyone know whether there is some sort of time zone option? I've
emailed Aastra who didn't come back to me. I would like to set the time
zone - e.g. Britain-London, in the cfg files so I don't have to set it
on 40 phones...
time
Just a contribution coming from an Asterisk-Newbie ignorant
Couldn't this behaviuor (The fake 2 phones, with the same ext #), be
achieved via a gruop configuration?
At least, in my [EMAIL PROTECTED], you can configure a group pointing to 2
different
extensions.
Regards.
Juan.
On 12/19/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Matthew wrote:
For the uninitiated among us (myself included) what is ACD login/logout
support?
The Polycom phones can send XML NOTIFY messages to signal to the server
the agent is logged in/out/paused. I know of no documentation on
Thanks, so would I be correct in assuming
time zone name: UK-London
time zone code: GMT
time zone minutes: 0
And will this have any affect on the daylight savings in march?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent:
I'm investigating an application which the client says uses an E1 trunk
with 30 voice channels and a D channel on 16 as normal, but without any
call signalling on the D channel.
In other words, as soon as I originate an outgoing call to a Zap channel
on the E1, the call immediately succeeds (is
Rich Adamson wrote:
If you take the word dynamic out of that, then can he effectively
have primary/secondary/backup systems that allows the user to
re-register and/or redial his call on a different * server?
I don't understand the question.
'dynamic' is used for registrations; if the peer
Matt Riddell wrote:
Morel Mosolff wrote:
Dear friends and business associates,
I will be out of office until January the 12th, 2006.
With kind regards,
Morel Mosolff
H1 more of these and I will start a loop on a spare high bandwidth
server :)
This person has been unsubscribed.
(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql,
terminating on one TE410
Mem: 3105772k total, 733928k used, 2371844k free,8k buffers
Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle
load average: 0.37, 0.39, 0.41
So that is ~80 calls per GB of
Still no joy, if I set my phone to a different time zone then reboot it
isn't being updated to use London.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pete
Barnwell
Sent: 03 January 2006 14:30
To: Asterisk Users Mailing List - Non-Commercial
Tony Mountifield wrote:
In other words, as soon as I originate an outgoing call to a Zap channel
on the E1, the call immediately succeeds (is considered answered) and
passes audio in and out on the specified channel. Obviously there will be
no such thing as an incoming call or a remotely
Hi,
I have about 53 SPA-2002 units out in the field. I've seen two or
three of them, now, exhibit an odd happening.
Users plug their phones into LINE1 (unless they have two lines). The
two users I've had issues with are both employees here who are fairly
knowledgeable in computers. They both
On Tuesday 03 January 2006 15:37, Noah Swint wrote:
Do you have a url for the device?
http://cyber-telecom.net/store/index.php?cPath=1
From: [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List -
On Tue, Jan 03, 2006 at 02:38:15PM -, Lee Archer wrote:
Thanks, so would I be correct in assuming
time zone name: UK-London
time zone code: GMT
time zone minutes: 0
And will this have any affect on the daylight savings in march?
Those are part of the definition of the timezone.
I had a problem which I spoke to Grandstream about. It seemed that
around 7 seconds in it goes for time sync and if it fails it doesn't
retry. This problem was highlighted by the .12 firmware and a Windows
DHCP server we were using. Upon moving to a Linux DHCP server the
process was much
Actually it worked, but only after I defaulted all the settings on the
phone and let it pick the config up fresh.
Anyone know if there is any headset config options to default to
headset/speaker?
Thanks
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
vmail*CLI realtime mysql status
Jan 3 10:14:20 ERROR[13666]: res_config_mysql.c:623
mysql_reconnect: MySQL RealTime: Failed to reconnect. Check debug for
more info.
vmail*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 2 days, 17 hours, 15 minutes,
Could be they pressed the DND code for the SPA (don't remember by
heart what it is, something like *xx). The easiest way to check is to
log into the http server of the SPA and check the status on the first
page.
On 1/3/06, Matt [EMAIL PROTECTED] wrote:
Hi,
I have about 53 SPA-2002 units out in
9 more days to go.
On 1/3/06, Morel Mosolff [EMAIL PROTECTED] wrote:
Dear friends and business associates,
I will be out of office until January the 12th, 2006.
With kind regards,
Morel Mosolff
___
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Hi ,
thank you for your answer,
If Prefix() command is only in CVS-HEAD, then how can you prepend leading
digits in a stable version?
It does not make any sense not to have this feature in a version downloaded
from thw Digium FTP site...
Budoka.
Use meetme appCheers, Giovanni Miano2006/1/3, Henry Margies [EMAIL PROTECTED]:
Hello,I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three WayCalling with a SIP or analog Phone is working perfectly.But if I try to do Three Way Calling with my ISDN Phone I get an error
message:
Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.
-Kerry
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To
Does anyone know anything about this company?
http://www.affinityvoiptelecom.com/
They claim to offer 911 routing and PS/ALI updates, etc.
___
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To UNSUBSCRIBE or update
That's possible, and I didn't think about that.(I wil check). however
I did totally wipe the configuration on the device *** RESET and then
reprogrammed it and the same problem happened, so I kind of doubt that
was the issue.
On 1/3/06, C F [EMAIL PROTECTED] wrote:
Could be they pressed the DND
From:
http://www.iaxtel.com/
The IAXTel Server is currently under maintenance. Some technical
difficulties, such as connection timeouts, registration timeouts, and the
inability to make phone calls may be experienced. Thank you for your
patience.
:(
b
-Original Message-
From:
Hi Antonio. h323 support is composed from several versions and
packages, including compatibility between asterisk versions and
asterisk-oh323 is important. I guess more people will be able to help
you if you privide more info.
Kind Regards
On 1/3/06, Antonio Marquez [EMAIL PROTECTED] wrote:
I
I agree, I liked the old ringtone 2 also (just abeep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed.
On 1/2/06, Remco Barende [EMAIL PROTECTED] wrote:
Hi Usman,Thanks for the
Just do:
exten = _12xx,2,Dial(${TRUNK}/0${EXTEN}|30,r) ; adding zero
exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r) ; not adding zero
The zero is added before ${EXTEN}.
I have only ever used the stable versions and have always done it this way.
--
--
Steven
May you have the peace and freedom
That message has been there for months.
On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote:
From:
http://www.iaxtel.com/
The IAXTel Server is currently under maintenance. Some technical
difficulties, such as connection timeouts, registration timeouts, and the
inability to make phone calls
I cannot get the following to work in an AEL script on 1.2.1
Dial(mynumber,timeout,M(mymacro))
Does anyone know if the Macro construction used above is supported in AEL?
or should I use
Dial(mynumber,timeout,mymacro)
John Melody
SyberNet Ltd.
Galway Business Park,
Yeah, saw that, and it had said that for like six months if I recall. You
would figure that since Digium features IAXTEL phone numbers so prominently,
that it would be a service that was actually capable of connecting to them.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
If the ata is sending busy here sip response back to asterisk it IS
most likely a DND or other call redirect setting that was user
programmed at the ATA.
I have seen the Linksys/sipura ATA retain USER settings when ADMIN
settings are reset to default with certain firmware versions.
I know, this is the sad part :(
b
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Tuesday, January 03, 2006 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXTEL??
That message has been
Hi
I wish to install asterisk 1.2 (the latest tar.gz from the site not the
CVS version) on an HP box with a TE110P (single port E1/T1)
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is to
set this dialing rule up.
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion
Kerry Garrison wrote:
Is IAXTEL still around? I
Asterisk-Users,
Is anyone out there using the SPA-1001 for integrating existing analog
phones into a VoIP setup? My question has to do with the MWI. From the
datasheet it says that it provides MWI Tones, and then that it provides
Visual MWL via FSK. What does via FSK mean? My exsting phone has an
Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.
That hasn't worked for many many months.
Much easier to reach digium by using the Demo that is/was installed in
all asterisk installs. When
IMHO use FC4.
Also after the install of the OS and all the required packages do a 'yum
update'.
Bogdan Moldovan
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary
For those bemoaning the lack of IAXTEL and wanting to contact Digium
what's wrong with:-
exten = ${DIGIUM},1,Dial(IAX2/[EMAIL PROTECTED])
worked 2 minutes ago.
--
Dave Cotton [EMAIL PROTECTED]
___
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Kevin P. Fleming wrote:
Rich Adamson wrote:
If you take the word dynamic out of that, then can he effectively
have primary/secondary/backup systems that allows the user to
re-register and/or redial his call on a different * server?
I don't understand the question.
I don't know if it was
It sounds like it might be dialplan instead of PRI related.
-bill
On 2-Jan-06, at 10:43 AM, Kristian Larsson wrote:
On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote:
On Mon, 2 Jan 2006, Kristian Larsson wrote:
I have an Avaya IP Office PBX connected to an
Asterisk system
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box without any
configuration. So should be good alternatively of phonecell or nokia pbx
etc..
Units are
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
happy new year, evrething work now :-D
the error came from France Telecom
thanks everybody
- --
Stephane Plichon | HASGARD
tel: +33 (0)472529881
fax: +33 (0)472177764
web: http://www.hasgard.net
email: [EMAIL PROTECTED]
jabber: [EMAIL PROTECTED]
~
By hints do you mean comments??
Seems a very odd solution, but I'm willing to give anything a go.
Regards
Andrew Gough
Senior Partner
GCD Technologies
Unit 414
Lisburn Enterprise Park
Ballinderry Road
Lisburn
Co Antrim
BT28 2BP
E: [EMAIL PROTECTED]
W: www.gcdtech.com
T: 028 9264 1144
Brett, Gary wrote:
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2
Try Fedora Core 4 (FC4). Works great.
___
We've been asked to quote for a large cluster running Asterisk and our
ITSP in a box product. The system will be SIP throughout, with mixed
codecs.
We're considering using Dell blade servers, 1855 or similar, on the
grounds that we normally use Dell machines and they work well, but we
need
Did you try running * under gdb? When it crashes, do a bt to get a
back trace and post it to the mailing list.
e.g.
% gdb /usr/sbin/asterisk
GNU gdb Red Hat Linux (6.3.0.0-1.84rh)
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License,
John Melody wrote:
I cannot get the following to work in an AEL script on 1.2.1
Dial(mynumber,timeout,M(mymacro))
AEL does not affect the syntax of arguments passed to applications, so
if this does not work then it is a bug in the AEL parser.
Mike Fedyk wrote:
Think of this scenario: You have two * RT servers running heartbeat and
one goes down. If the SIP registration information was kept in the DB
tables, the backup server could take over the ethernet and IP addresses
and continue without forcing the phones to re-register.
Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a PSTN
will I have answer supervision so that my script won't initiate too early?
Jason Wolfe
[EMAIL PROTECTED]
c (770) 561-6956
This e-mail transmission may contain information that is proprietary,
privileged and/or
Hi all,
I am trying to get DTMF digits from X-pro, through a grandstream ATA, to a FXS to FXO converter for outgoingPSTN calls. I could hear second dial-tone from the phone line connecting to the converter. However, no PSTN dialing occured after DTMF digits was sent from X-pro.I tried while
One thing to be aware of is that Dell blade (as well as many other brand)
servers are very heavy beasts.
In any deployment with these, check the physical dimensions, check the
weight and ensure that it will actually install into the rack that you are
using. Also, check the power consumption
Any thoughts on CentOS-4.2?
It is based on RHEL4 update2.
It has the 2.6 Kernel.
I am currently using CentOS-3.5, which is based on RHEL3 update5, with no
issues. The Kernel is 2.4.21-32.0.1.ELsmp.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
Just as an update, as of this morning, the Techs at Digium do have this
working properly and are in the process of trying to determine if the reason
mine is not working properly is due to a hardware or software problem with
the card.
Kerry Garrison
Director of Technical Services
Tech Data Pros -
Haven't seen a post
to this list since last night. Don't know if there'sa problem or
not.
I'm trying to record
calls for SPECFIC agents, which queues.conf and agents.conf don't seem to
support. Someone suggested I just put a monitor() command before the Dial() so
that when the Queue dials
Kevin P. Fleming wrote:
Mike Fedyk wrote:
Think of this scenario: You have two * RT servers running heartbeat
and one goes down. If the SIP registration information was kept in
the DB tables, the backup server could take over the ethernet and IP
addresses and continue without forcing the
On Sat, Dec 31, 2005 at 10:05:19AM +0100, Olle E Johansson wrote:
We're currently planning a new generation of chan_sip that will have a
different authentication scheme, not based on the from: header unless
it's a local policy to require the From: header to be the same as the
Digest auth
Brett, Gary wrote:
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core 3, and are there any install guides out there that are
recent enough for asterisk 1.2
Use Debian or Centos (Free RHEL).
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Jason D. Wolfe a écrit :
Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a PSTN
will I have answer supervision so that my script won't initiate too early?
I'm not sure to understand you. If you don't use Answer() before you use
Dial(), asterisk won't answer
I generally use CentOS. Haven't tried CentOS 4 with Asterisk yet, but I'm
sure it'd work fine.
It's generally less of a moving target than Fedora is as far as updates
are concerned. CentOS 3.x will get updates as long as Red Hat is providing
them whereas FC1 servers and FC2 servers we set up a
I'm currently using CentOS 4.2 in my home install on a P3-600/512MB/40GB
HDD with a X100P clone and it works great. Using Asterisk 1.2.1.
Ryan
Any thoughts on CentOS-4.2?
It is based on RHEL4 update2.
It has the 2.6 Kernel.
I am currently using CentOS-3.5, which is based on RHEL3 update5,
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