Hi,
In my application I want to have the
sysdate + 5 minutes.
I know that the sysdate is in the variable
${DATTIME}
But now I want to now how I get the
sysdate + 5 minutes into a variable?
Does anybody knows the answer?
Kind Regards
Arjan Kroon
The lockup is related to zaptel, i promise :)It's been a while, but I seem to remember Zaptel having some strange interactions with USB in 2.4 kernels. I'd strongly recommend 2.6
On 4/20/06, T.S [EMAIL PROTECTED] wrote:
Hello folx!I just started to play with *. I first installed it this past
I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA
186 and fax pass thru well. I have tried to establish modem connection (from
computer connected to ATA = SIP = * = E1 = Telco = pstn = another modem)
and I do connect (at 14,400) but connection end after a minute.
${EPOCH} * Current unix style epoch
Add your 5mins as seconds, and convert if necessary, you could do it
like this in the dialplan to give the same format as ${DATETIME} (which
is deprecated by the way):
${STRFTIME($[${EPOCH} + 300],,%d%m%Y-%H:%M:%S)}
Read doc/README.variables to find out
Tomislav Parčina wrote:
I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA 186 and fax pass
thru well. I have tried to establish modem connection (from computer connected to ATA = SIP
= * = E1 = Telco = pstn = another modem) and I do connect (at 14,400) but
Yeah, I got it a couple of times.
Doug Lytle wrote:
An outside caller started to leave voice mail.
The CLI shows:
Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: gsm,
0x8295d40
-- x=1, open writing:
On 13:40, Thu 20 Apr 06, Douglas Garstang wrote:
Does AMP also let you split up each charity so that each only has access to
manage their own content? That seems to me to be a pretty big limitation of
all the Asterisk management software out there. It's designed to be used by
one company to
Which USB Phones, come with G729 support?
I am looking for one which has the G729 in the software installed on disk
itself, so that if the users can use onscreen dialling with headphones if they
want.
/Obelix
___
--Bandwidth and Colocation provided
Grab the UNIX timestamp and add 5*60 to it.
In my application I want to have the sysdate + 5 minutes.
I know that the sysdate is in the variable ${DATTIME}
But now I want to now how I get the sysdate + 5 minutes into a variable?
Doe's anybody knows the answer?
Hi all
I install sphinx2 successfully but when i execute
sphinx2-server, i have the error below:
ad_oss.c(105): Failed to open audio device(/dev/dsp):
No such device
FATAL_ERROR: server.c, line 476: ad_open() failed
What's the matter?
I also want to know how i can do in Asterisk to use
On 04/21/06 05:35 Sean Kennedy said the following:
I have a vague memory of reading about this somewhere, but searched @
the wiki AND through google aren't turning up anything useful.
take a look at http://bugs.digium.com/view.php?id=6897
there's a patch there for 1.2 with another for trunk
Have a look at the Dial command,
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
the w and W option allow you to start recording at any time with the *1
keypress.
Is there a way to record a call conversation starting in the middle of
the call? I know I can recording whole
Hi!
I have some (short) question about the Eicon DIVA (V-)4BRI cards I want
to have (short) answered before buying the DIVA card. I know there are
several Eicon guys active on the list, thus I ask on the list instead of
directly to Eicon so that all other will benefit as well.
1. Do Eicon
Hi
All
I
want bulding next scenario conferences:
Dial
number conference roomafter automatic connect 2
users
Can
any one help me with samples this scenario
Thanks
Viktor
Tatianin
___
--Bandwidth and Colocation
On 20 Apr 2006, at 16:09, Peter Beckman wrote:
I've tried Idefisk and Cubix Softphones, and they both work fine,
except
for two issues:
1. Idefisk seems to have a longer delay between the time I can hit
tones, and
2. Cubix, while can send DTMF faster, never actually connects
On 20 Apr 2006, at 16:39, Robert Rozman wrote:
- Original Message - From: Time Bandit
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 20, 2006 4:18 PM
Subject: Re: [Asterisk-Users] Jingle support
Hello Klaus,
I will answer your questions In turn:
1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
Yes they do.
2. Can the clock (master/slave) be configured independent from the mode
(TE/NT)
No
3. Is the PIN layout for TE or NT mode?
The PIN layout is for TE mode
4. When I change
Ive had Asterisk working on a test platform
really well, but Ive never found a decent web front end, that works in real-time.
Ive got a couple of incoming numbers that Id
like to have some IVR on (i.e. select this option etc), and then distribute the
calls appropriately to various SIP
That only works if the default password is still cisco.. in this case it was
not.
--
~Shaun
Joseph Rothstein [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
You can usually unlock the phone and then erase the config using the
setting
sbutton. Push the setting button, nafigate
On Fri, April 21, 2006 11:45, David Waugh said:
Hello Klaus,
I will answer your questions In turn:
1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
Yes they do.
2. Can the clock (master/slave) be configured independent from the mode
(TE/NT)
No
Thus, what is the limitation?
TE =
turns out the g option for the macro when using dial doesnt allow both ends
to be hung up with setting MACRO_RESULT=ABORT..
My workaround was a bit ghetto, setup a context called hangup-caller and
Set(MACRO_RESULT=GOTO:hangup-caller^s^1)
jsut fyi..
--
~Shaun
Shaun [EMAIL PROTECTED] wrote
End result was i gave in for a annoying setup... put the phone and a server
on a network by it self, setup a dhcp server with a tftp address and flashed
the phone to 8.2 which also reset the password back to cisco
--
~Shaun
Shaun [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hello All,
I'm facing problems with Unicall on this scenario :
CentOS 4.3 - Running on x86_64
Asterisk 1.2.7.1
Zaptel 1.2.5
When running zttool , shows all Spans OK.
But I can't receive and make calls.
I tried to change many parameters and still doesn't work.
Any clues ?
* unicall.conf
On Friday 21 April 2006 00:28, Kerry Garrison wrote:
Didn't help. Could I be missing something else?
In Avi's footsteps, here is my phone.cfg and sip.conf entry. This works for
12 phones. Note that I'm not subscribing to anything on the Polycom;
Asterisk sends MWI for the mailbox to the
On Fri, 21 Apr 2006, Klaus Darilion wrote:
Hi!
I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to
have (short) answered before buying the DIVA card. I know there are several
Eicon guys active on the list, thus I ask on the list instead of directly to
Eicon so that
Hello list users, I come to you in order to ask you
your best recommendation for a large scale production Server, which Hill be
your best recommendation between [EMAIL PROTECTED] and an installation from scratch
with Fedora Core 4 and asterisk compiled by sources??
Thanks in advance,
On Fri, 21 Apr 2006, Klaus Darilion wrote:
On Fri, April 21, 2006 11:45, David Waugh said:
6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
is the better (more expensive) card which also offers FAX on/offramp.
Nevertheless I can use V-4BRI for faxing when using
Hello Klaus,
Normally, Diva Server adapters are operated as terminal equipment. In
this case, they derive their timing from the signal received from the
NT, for example PSTN or PBX, and use this derived timing to synchronize
their transmitted signal. If you use the Diva Server adapter as network
Hi all,
having performance problems with various SIP-Phones, the manufacturer
adviced us to add these parameters in sip.conf - unfortunately, neither
one of us has an idea what these are supposed to do.
I've seen various configuration files (sip.conf, iax.conf) posted on the
net or this list
Install from sources. Run only Asterisk on the production machine, no extra
processes. If you must have databases, GUI, AGI and ... put those on another
machine and call them across the network. Also, I am not sure how much it
helps, but eliminate asterisk modules you will not use by using
Hi,
I'm planning to install a new Asterisk server with a Digium TE410P card.
Can I use Red Hat Advanced Server 4 (latest update)?
Is this a good choice?
Is recompiling Asterisk simple with kernel 2.6?
Thanks
--
Domenico Viggiani
___
--Bandwidth and
Hi,
I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
It works fine when I have just 1 voice installed. Now I have 2 voices in
the same language installed but I can't seem to find the way to select
which voice to use in Swift's application in Asterisk. Does anyone know??
Red Hat AS 4 is the same as CentOS4X (with the Red Hat references stripped) and
Asterisk works just fine on it.
-Original Message-
From: Mimmus [mailto:[EMAIL PROTECTED]
Sent: Fri 4/21/2006 6:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Type swift at the command line so you can see the -options. Then modify the
line to use the correct switch and specify the name of the voice you want to
use.
Thanks,
Steve
-Original Message-
From: Pimjai Wesnarat [mailto:[EMAIL PROTECTED]
Sent: Fri 4/21/2006
Cubix has always crashed on me while using moderately. Nice looking phone but
not stable.
Idefisk works great.
On 20 Apr 2006, at 16:09, Peter Beckman wrote:
I've tried Idefisk and Cubix Softphones, and they both work fine,
except
for two issues:
Did you get an answer to this? I am interested in SIP to SIP calls on other networks thereby by-passing the pstn.
-- Original message -- From: Nick Hoffman [EMAIL PROTECTED] Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? For instance, an
On Fri, April 21, 2006 12:33, David Waugh said:
Hello Klaus,
...
I haven't tried spandsp and a V-Series card as the card was not designed
for fax applications. As a result it has never been tested. If you want
to fax with the card then we would recommend a normal Diva Server card.
It would
Hi allHas anyone seen this kind of messages : Apr 21 12:04:12 NOTICE[89928]: chan_sip.c:3449 process_sdp: Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp'I get this using a priorietary (Airspan's prime, ex-arelnet) E1 gateway with asterisk. It seems like the SIP
Hi Armin!
Thanks for the detailed answers and isdn for Linux basics. I will take
the opportunity to ask some more questions :-)
On Fri, April 21, 2006 12:24, Armin Schindler said:
On Fri, 21 Apr 2006, Klaus Darilion wrote:
3. Is the PIN layout for TE or NT mode?
It is TE PIN layout, you
James Nunnerley wrote:
I’ve had Asterisk working on a test platform really well, but I’ve never
found a decent web front end, that works in real-time.
I’ve got a couple of incoming numbers that I’d like to have some IVR on
(i.e. select this option etc), and then distribute the calls
Hi!
I've forgotten to ask an important question:
Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?
thanks
klaus
btw: we should collect QA somewhere on a Wiki.
On Fri, April 21, 2006 12:33, David Waugh said:
Hello Klaus,
Normally, Diva Server adapters are operated as
Ohh yeah good point. I had a similar issue when I started using FreePBX
and it didn't fill out the mailbox field automatically. Once I added
the [EMAIL PROTECTED] there the MWI started working as well.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On Fri, 21 Apr 2006, Klaus Darilion wrote:
On Fri, April 21, 2006 12:33, David Waugh said:
Hello Klaus,
...
I haven't tried spandsp and a V-Series card as the card was not designed
for fax applications. As a result it has never been tested. If you want
to fax with the card then we would
Armin Schindler wrote:
Using spandsp and V-4BRI does not work?
That will work. It's just that the on-board fax capabilities won't work, but
any other software fax will work like with other cards.
Just a note that I've never managed to get this to work on my V-4BRI
cards: If I attempt to
On Fri, 21 Apr 2006, Klaus Darilion wrote:
Hi!
I've forgotten to ask an important question:
Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?
Yes, and each port can be configured separately.
thanks
klaus
btw: we should collect QA somewhere on a Wiki.
Yes. For
On Fri, 21 Apr 2006, Avi Miller wrote:
Armin Schindler wrote:
Using spandsp and V-4BRI does not work?
That will work. It's just that the on-board fax capabilities won't work,
but any other software fax will work like with other cards.
Just a note that I've never managed to get this
Quoting Pimjai Wesnarat [EMAIL PROTECTED]:
Hi,
I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
It works fine when I have just 1 voice installed. Now I have 2 voices in
the same language installed but I can't seem to find the way to select
which voice to use in
Hi Klaus,
Thanks for the detailed answers and isdn for Linux basics. I will take
the opportunity to ask some more questions :-)
On Fri, April 21, 2006 12:24, Armin Schindler said:
On Fri, 21 Apr 2006, Klaus Darilion wrote:
3. Is the PIN layout for TE or NT mode?
It is TE PIN layout,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
By luck, maybe. The only solution that looks like it should be fairly
solid is V.150, and I've only seen that on Cisco boxes so far.
Hi Steve!
Can you tell me more about Cisco box that are you talking about?
--
Tomislav Parčina
Lama
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
is there any way to make the Flash Operator Panel show which agents are
logged in in a specific queue? (both static and dynamic agents)
I've played around with the queue / queue agents settings from the Flash
Panel documentation
Thanks a ton!!
When using Extensions mode (the default) this would be:
[EMAIL PROTECTED]
When Using Users and Devices mode this would be:
[EMAIL PROTECTED]
Thanks for the guidance there, this has been driving me nuts.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
Try specifing [EMAIL PROTECTED] I know their have been some changes
with the implicit defining of the voicemail groupsthat may have
something to do with it... I didn't have to do anything special for my
polycoms.
Sean
On Fri, 2006-04-21 at 06:17 -0400, Andrew Kohlsmith wrote:
On Friday 21
A couple of weeks ago, libmfcr2 has a small error in the tone
signaling for the call setup, that was fixed 2 weeks ago or so,
please, wich version of libmfcr2 are you using? if you dont know try
upgrading to the latest version. Im pretty much sure that you have the
very same problem we had.
Hello,
I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
jumper set (i.e closed to use the E1 facility.)
but when i connect the E1 from my telco the LED on the TDM card is green
and also when i look in zttool the status are ok.
when i try to place a call out I get
Try pri show span 1 and send me the result.
Augustine Olaifa wrote:
Hello,
I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have
my jumper set (i.e closed to use the E1 facility.)
but when i connect the E1 from my telco the LED on the TDM card is
green and also when i
Hi listers,
I am looking for people who have used Power over Ethernet switches,
primarily in conjunction with Polycom IP 501's. I've been looking at the
Linksys SRW224P, since I've had good luck with the SRW224 in our office.
However, Nortel, Cisco, Adtran, etc. all have an offering, all
D-link has a nice one, optional 5 year warranty on some of the
commercial stuff
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi listers,
I am looking for people who have used Power over Ethernet switches,
primarily in conjunction with Polycom IP 501's. I've been looking
On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:
D-link has a nice one, optional 5 year warranty on some of the
commercial stuff
Though beware, some of the D-Link ones only have half the ports with
PoE.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612
Netgear makes a 24 port Layer 3 Managed Switch, with PoE on all 24 ports. It supports both IEEE 802.3af PoE as well as the proprietary Cisco PoE scheme (although the support for Cisco PoE is undocumented). Got one about a year ago for around $1000, which isn't too shaby for a a switch that can do
Why do you comment these lines:
;channel=31-45
;channel=47-61
and put those in the zapata.conf?
bchan=31-45,47-60
dchan=46
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hey everyone,
Hopefully I can describe the problem well enough so bear with me.
There are 3 companies that are tied into our asterisk server. Company A
(us) uses the default settings for music on hold. Companies B and C
however, want something different. For them I have when a call comes
Hi All,
I'm trying to add this function to my find-me application: when all
available numbers are dialed in parallel , if any number is busy, take
it at busy and go to voice mail. I read the Dial() Application but
there's nothing written about this. My question is, is it possible to do
this
Hello,
I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
jumper set (i.e closed to use the E1 facility.)
Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were
completely obsolete now?
___
--Bandwidth and
Jumpers must still be on for E1 mode.RobOn 21/04/06, Remco Barende [EMAIL PROTECTED] wrote:
Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
jumper set (i.e closed to use the E1 facility.)Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers
Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.
There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.
Any help appreciated.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
Hopefully I'm not just missing some little detail here. We're trying to
set the HANGUPCAUSE on SIP channels to have our softswitch play the proper
recording instead of answering the call on Asterisk to play the message.
It appears that no matter what the HANGUPCAUSE is set to, Asterisk always
I have set up an operator queue for our receptionist.
That way, if she takes a break or is out, by logging out of the queue,
calls to the Operator can be handled by other agents.
I have set strategy = roundrobin in queues.conf.
According to the book ATFoT, roundrobin always starts with the first
Hello to all
Can someone tell me if its possible to implement a SIP domain with
Asterisk (im trying with [EMAIL PROTECTED]).
With a SIP domain I mean:
-users having URIs with [EMAIL PROTECTED] ( instead of [EMAIL PROTECTED] )
-being able to reach our users anywhere in the world with SIP URIs
Hi Jim,
The function roundrobin makes with that asterisk directs the calls for the next free agent, but notorderly. I use the same strategy and functions here very well. Thedifference is that only use the functions ofagent loginokand agent loginoff.
I wait to have helped
Good luck
Regads
Josué
-Original Message-
From: Wasif [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 20, 2006 4:25 PM
To: 'asterisk-users@lists.digium.com'
Subject: Asterisk FAx-to-Email
Hi,
I get error when my DID hit to asterisk box which I am using for FAX to
Email Service. Sometimes Fax goes through
Hi,
Check the script. You can assign the voice by -n option, e.g.,
/opt/swift/bin/swift -n Diane
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane Young
Sent: Friday, April 21, 2006 9:17 PM
To: Asterisk Users Mailing List -
Weird, I just received a new TE210P card (should be identical only 3.3v)
but I cannot find any info on jumper settings om the Digium site?
But then again the installation info on the Digium site really sucks.
On Fri, 21 Apr 2006, Rob Lith wrote:
Jumpers must still be on for E1 mode.
Rob
Hi,
How can we change the FROM address when Asterisk sends mail (in FAX-to-Email
feature). For example it is sending [EMAIL PROTECTED] in FROM
address; I need to change it to [EMAIL PROTECTED]
Any help?
Wazb
___
--Bandwidth and Colocation provided
how about, in sip.conf, [EMAIL PROTECTED] in
the [section] for that device?
Bill Gibbs wrote:
Put your voicemailbox number (usually extension) in the 1.subscribe field.
Bill
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 4/20/2006 7:32 PM
Peter Fern wrote:
Probably because the Local proxy channel drops out once the two sides
have been bridged. If you want the Local chan to stay up, use the /n
parameter and the local channel won't perform the native transfer. This
does have it's own problems, but should do what you want.
Wai Wu wrote:
I notice those options. However, I was looking to start the recording
through a third party control program. I know I can do this via
chanspy, but is there better way?
Not that I know of... I was looking for something kind of similar, and
ended up actually using a conference, and
To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ?Shall we then allocate destination numbers and or ports for each of those 2 applications ?
And if you want to offer to every user, a unique extension for fax and voice, would
SMC 6824MPE.. Does 24 ports POE, with 2x 1GB uplinks, (RJ45 or GBIC)
L3 Managed switch.
We've got four of them here, and I think they're great, the cost was
really reasonable.
Ingram no longer lists the MPE model, but it should be available still.
Chad
-Original Message-
Hi listers,
Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ?For instance:
Location 1:- 200 users on a legacy PBX- among those users, 50 have access to
Anyone know how to set the wellgate unit so incoming calls
pass on directly to asterisk?
Right now incoming calls ring twice and I hear a recording saying enter
the extension. If I go enter the extension it goes on to asterisk just fine.
I just want the incoming call to go directly onto
It really depends on the PBX in use. Avaya uses DTMF tones to light
the MWI, you can find examples on the wiki on how to do it. In which
case you shouldn't have any problem doing it.
Most of the bigger phone systems I have worked with allow the same
thru simple DTMF tones.
On 4/21/06, Olivier
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi listers,I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's.I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office.However, Nortel,
Hello list,
we are testing an easier way to install QueueMetrics on an [EMAIL PROTECTED]
box (or any other CentOS/RHEL) using the yum package manager.
This is still experimental, so it may as well work as not work.
We are looking for testers who are willing to try this at home and any
On 21 Apr 2006, at 18:21, Olivier Krief wrote:
To benefit from DIVA Server 4BRI fax hardware capabilities, what is
the best software combination ? Asterisk and Hylafax ?
Shall we then allocate destination numbers and or ports for each of
those 2 applications ?
And if you want to offer
This is coming from an * noob. :)
I've got two customers, they both are replacing their phone systems with
VOIP, and we need to retain both their existing dialplans.
One has 5 extensions starting at 100, and the other has 10 extensions,
starting at 100.
Is there a way to have the same
On 4/21/06, Rick Smith [EMAIL PROTECTED] wrote:
This is coming from an * noob. :)
I've got two customers, they both are replacing their phone systems with
VOIP, and we need to retain both their existing dialplans.
One has 5 extensions starting at 100, and the other has 10 extensions,
- Original Message -
From: Tim Panton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 21, 2006 11:43 AM
Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ?
On 20 Apr 2006, at
that feature does not exists AFAIK, but you can request it in
bugs.digium.com, or offer some money to someone to include it for you.
Regards
On 4/21/06, Pimjai Wesnarat [EMAIL PROTECTED] wrote:
Hi All,
I'm trying to add this function to my find-me application: when all
available numbers are
My favorite one is this one.
http://www.provantage.com/d-link-systems-des-1526~7DLNS046.htm
On 4/21/06, Steve Kennedy [EMAIL PROTECTED] wrote:
On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:
D-link has a nice one, optional 5 year warranty on some of the
commercial stuff
Hello,
I have a problem with one of three [topic] phones. The phone, which is
on the LAN in the same subnet as Asterisk, keeps unregistering from the
Asterisk server. Whan it is unregistered there is no way to make a phone
call from it, but once it is rang by any other of the phones it
registers
I tried by just upgrading to Ast1.2.4 but same problem.
Then I tried to install OH323 but I have this error when compiling :S
chan_oh323.c: In function `reload_config':
chan_oh323.c:4677: warning: implicit declaration of function `sscanf'
chan_oh323.c: At top level:
chan_oh323.c:3244: warning:
We're using Cisco Catalyst 3560 Series 48 port PoE switches. So far,
*they just work*.
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___
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Asterisk-Users
On Fri, 21 Apr 2006, Olivier Krief wrote:
To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best
software combination ? Asterisk and Hylafax ?
You can use any combination of CAPI based software in parallel. You just
need to create the rules for which application shall act
I had a similar problem with a GS101, although with mine, I could make
OUTBOUND calls from the phone, but because it wasn't registered, it
wouldn't ring if called. I don't know the exact solution, but two
things I did was to tell it NOT to subscribe to MWI in the GS config
itself, and second, I
Hi Bart,
If it's anything like the problem we had, you are probably getting what
sounds like screeching noises during MOH playback? We had this problem
and made it go away by turning off hyperthreading in the server BIOS
and starting Linux with noht - this was on a dual Xeon machine.
Hope
Hi Domenico,
We're using RHEL 4 ES with no obvious issues
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Apr 21, 2006, at 3:59 AM, Mimmus wrote:
Hi,
I'm planning to install a new Asterisk
hello
I will buy a server to make an IVR solution with
asterisk and a te110p T1/E1 digium card.
I have two options:
1/ HP Proliant ML370 G4 : Xeon 64bits 3,2Ghz, 1Go
Ram, 3 disks SCSI 73Go
2/ Dell PowerEdge 2800: Xeon 64bits 3Ghz, 1Go Ram,
3 disks SCSI 73Go
I use linux fedora core 3 and I
Is there a way to have an extension match on a sip address? I've tried
the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no
good.
Is there a better way?
--
Jon-o Addleman - http://redowl.dyndns.org
___
--Bandwidth and
Hey guys,
I'm actively trying to get the big picture on how all this works and
relates to each other.
I've gone through some basic examples from the book and from the sample
files just fine.
Now, I've setup an account with a VOIP provider which does IAX termination
(exgn.net)
After getting an
Steve Kennedy wrote:
Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.
There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.
Any help appreciated.
Steve
Steve,
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