> "Kris" == Kris Stark <[EMAIL PROTECTED]> writes:
Kris> On a different note - is something up with the freenum.org enum
Kris> lookups? ... I've had them fail on all US numbers...
The nameservers for freenum.org. have glue records for 1.freenum.org.
that point to garthim.fox-den.com. (which i
Matt Lawson wrote on Mon, 09 February 2004 16:55]
>Hmm. Both Voicepulse and Nufone don't seem to be able to dial
>out 800 numbers. Are 800 numbers treated differently somehow?
No difficulty on this end to dial 800 numbers via Voicepulse... I just
don't do so normally - as others stated, I eithe
Great idea, James.
David, you could use the manager interface to automagically do all this.
When one person initiates a call to the speaker phone, the manager interface
could automatically send the speaker extension to the conference room.
Voila, you're both hooked up!
Check out mattf's thread, "
Today I had five system freezes. After re-examining the log file I see
the following line precedes each freeze:
"Got event 2 (Ring/Answered)"
However, I also get this same message numerous times without a freeze.
Many are followed by "Unable to create channel of type Zap".
This is a single pr
> > but I will when I can figure out how to do it securely without
> giving
> > anyone who dials my FWD number access across my Asterisk box to
> outbound
> > calls through Nufone.
I land inbound calls from service XXX in a context called
[inbound_XXX] examples are inbound_FWD and inbound_PSTN.
For thoses who are interested, Firefly 1.4 has just
been released. (http://www.virbiage.com/firefly/)
Of interest to asterisk users
- there's now the ability to save urls on the
contact list.
- asterisk's dial format is now supported (eg
iax2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) - Push
I was hoping for a SIP connection. No D to A and then A to D.
> Plug the Packet8 DTA310 into an X100P and go.
>
> On Mon, 2004-02-09 at 14:06, Jim Flagg wrote:
> > - Original Message -
> > From: "Eric Wieling"
> > To: <[EMAIL PROTECTED]>
> > Sent: Friday, January 23, 2004 11:07 AM
> > S
On Mon, 9 Feb 2004, Tim Petlock wrote:
[ Long Explanation Deleted ]
> Nufone and Voicepulse would have to maintain some number of trunks with
> an ILEC or CLEC to complete toll-free calls.
I dial 800 numbers all the time from my Nufone account without problem.
Hell, my DID through Nufone -IS-
> It didn't occur to me to add FWD to my config to complete such calls -
> but I will when I can figure out how to do it securely without giving
> anyone who dials my FWD number access across my Asterisk box to outbound
> calls through Nufone.
I actually use IAXTel for my 800 number service. I f
I was hoping to be able to avoid using the V1005 altogether. And have vonage
call the asterisk server. I imagine it would be better quality and
identification.
I have it working with the X101P
Jacques
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Youne
The business reason is that Voicepulse and Nufone probably have a
combination of other VOIP business relationships and direct T1's to long
distance resellers and those are used to complete calls.
I have never experienced a case where a direct T1 to a long distance
carrier could complete toll free
Hi - I'm wondering if anyone has had any experiences with this:
I've got an X100P card, and an HP DL380 Gen3 server I want to put
it in. The server has 2xPCI-X 100 and 1xPCI-X 133 hot-plug slots.
It's my understanding that the PCI-X slots are completely backward-
compatible with standard PCI card
On Mon, 9 Feb 2004, Steve Kennedy wrote:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
> Debian/Fedora Core ?
Debian works good. Packages for 0.7.1 are in the unstable tree and have
som
Has anyone tried to connect either X100P or the TDM400P cards in paralell to
phone lines and record incoming calls?
Best,
PauloHM
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Has anyone attempted to connect an NEC ITR-16D-2 phone to * ? I'm very
new to Asterisk, and have a couple of these IP phones to play with.
I can't find much technical info on the phones ny STFW, they may be
some proprietary items.
--
Jay Hennigan - CCIE #7880 - Network Administration - [EMAIL P
ipsec vpn's. cisco's pix 501 is cheap and works well for the home office
types.
That way the * box just appears to be on a different subnet as far as the
phone is concerned. nat'ing becomes a non-issue.
At least that's what we plan to do.
- Chris Clifton
- Original Message -
From: "To
This may be a way to do what you described with a $20
speaker phone and no phone modifications:
Using the speaker phone dial-in to a conference room on the
asterisk
Then whenever anyone wants to call this extension, you can
route their call to the conf room and they can have a two-way
conver
Dear all,
I am one of the people who answer the FWD welcome
line. Since I don't want my phone ringing at 2am I have the 5 number
routed to a call queue. Currently I have 2 extentions 271 which will log
my phone into the queue and 270 to log it out. What I want to know is...
Is there a
[I have found further info in ethereal down at the bottom of the note]
I have a dual homed asterisk box like this:
I have real and fake number on the same LAN, and the fireall is aware of
this:
FW/GW real IP eth0 (Filtered)
fake IP eth0:1 (NATTED)
-real IP et
Hi,
Anyone knows if there is a MGCP media gateway implementation for
* ? The stock chan_mgcp.c seems to be a call agent implementation
only and reflected in mgcp.conf as such.
PK
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, Februa
There is an auto-answer speakerphone that might do what you
described:
http://www.vikingelectronics.com/products/apartmententry/k-1700-3(rd).html
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
David
Schumann
To: [EMAIL PROTECTED]
Sent:
G'day Tomica,
On Tue, 10 Feb 2004, Tomica Crnek wrote:
> Are dual line devices in skinny mode (79xx, ATA) supported by Asterisk?
I have multiple line presentations on my SCCP 7960 using the chan_sccp
code. http://www.zozo.org.uk/pages.shtml?page=sccp will get you to one
place it can be found,
Yonah Wolf wrote:
All,
I am relatively new to Asterisk, and among the many things that I
would like to use it for, one of them is using it to get a Cisco 7910
to work. Does anyone know where I can get the Skinny or SCCP protocol
extensions for Asterisk?
Secondly, has anyone got this to work?
Matt Lawson wrote:
Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800
numbers. Are 800 numbers treated differently somehow? Or is there a
"business reason" for disallowing them? It makes the ringing sound
but never connects.
You can call toll-free numbers via NuFone.
What are you trying to accomplish? What is the architecture of the system
you are trying to get operational with NAT?
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL
From: "Glenn Dalgliesh" <[EMAIL PROTECTED]>
I am assuming the problem you are trying to solve is no audio. Are both =
the phone and asterisk on public ip address?
-
The problem is the ICMP messages in response to what presumably is an audio stream, as originally described.
Both devices ar
Hi Bill,
Your problem seems to be a codec negotiation issue, I think you need to
specify for each SIP peer:
disallow=all
allow=alaw
allow=ulaw; and any others that you might need
Paul
> - Original Message -
> From: Bill Michaelson
> To: [EMAIL PROTECTED]
> Sent: Tuesday, February 10
Are dual line
devices in skinny mode (79xx, ATA) supported by Asterisk?
Hi,
Here is the error I am getting when trying to run make for zaptel that
exits the make file:
zaptel.c: At top level:
zaptel.h:998: error: storage size of `confin' isn't known
zaptel.h:999: error: storage size of `confout' isn't known
zaptel.c:5897: error: storage size of `zt_fops' isn't known
If you read to the end, the guy is not having trouble if he connects to
the PSTN via the first digit. I would be concerned that either the DTMF
detection in asterisk then was too strict where the PSTN would otherwise
detect it. So while it is possible, it seems odd that Digium wouldn't
have noticed
Use fwd.pulver.com
On Mon, 9 Feb 2004, Matt Lawson wrote:
> Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800
> numbers. Are 800 numbers treated differently somehow? Or is there a
> "business reason" for disallowing them? It makes the ringing sound but
> never connects.
I am assuming the problem you are trying to solve
is no audio. Are both the phone and asterisk on public ip
address?
- Original Message -
From:
Bill Michaelson
To: [EMAIL PROTECTED]
Sent: Monday, February 09, 2004 5:26
PM
Subject: Re: [Asterisk-Users]
asterisk
Hi,
> -Original Message-
> > What I would do i this situation is work out a fax <-->
> email gateway.
> > Best case this could be done entirely with software on the asterisk
> > box, worst case a faxmodem hairpinned into an fxs card
> using hylafax.
>
> Why exactly would hylafax be a "
Arg.. my posting was mangled by text-wrapping. Sorry.
Here again...
sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[248379]
username=billdesk
type=friend
host=dynamic
canreinvite=no
mailbox
Nicolas Bougues wrote:
On Mon, Feb 09, 2004 at 04:36:41PM +0100, [EMAIL PROTECTED] wrote:
Some vendors sells phones with dual ethernet ports. Are these just
incorporating a hub/switch functionality? The reason for my question is
that the normal case for a DSL customer is the possibilty to use o
Hi
a gateway and an ATA186 for test.
1.when I set them up in H323 mode, put gateway's IP in ATA186's
configuration ,ATA186 could send calls directly to gateway, and they
worked just well.
2. after I changed them to sip mode, and ATA186 got no dialtone.
My question is:
In order to send ca
All,
I am relatively new to Asterisk, and among the many things that I would like
to use it for, one of them is using it to get a Cisco 7910 to work. Does
anyone know where I can get the Skinny or SCCP protocol extensions for
Asterisk?
Secondly, has anyone got this to work? Is this an incredul
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:9242(256) to H323:28967(8)
Feb 9 15:37
See below for a previous post of mine. It doesn't offer a "dial pad"
like most normal phones, but won't get stolen, offers the intercom
service you want, and has a big fat button for people to call a
"hotline".
JT
I've looked around and found previous discussion about this, but so
far I have
Tim Sailer said:
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered SIP to be seen
> as unavailable. Am I just missing something obvious, again?
>
> Ti
Right - OK - sans comments for brevity: sip.conf: [general] port = 5060
; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context =
default ; Default for incoming calls [248379] username=billdesk
type=friend host=dynamic canreinvite=no mailbox=1234 context=demo
extensions.conf: [general
Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800
numbers. Are 800 numbers treated differently somehow? Or is there a
"business reason" for disallowing them? It makes the ringing sound but
never connects.
___
Asterisk-Users
Ahh, I tried that but left out the 1 to prefix the area code. That did it,
thanks.
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Monday, February 09, 2004 3:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE:voiceglo sip
sorry, after some time and a couple of beers i found the solution by myself
...
In germany we have to set the callerid in national style like 72317810714
cause otherwise the telekom will change it again ...
Thanks for help,
Johannes
> -Ursprungliche Nachricht-
> Von: [EMAIL PROTECTED]
>
Your problem has the classic symptoms of using inband DTMF and a
compressed codec. If you are using a codec OTHER than ULAW or ALAW you
MUST set the DTMF mode on the PHONE AND in Asterisk to be rfc2833 (or
INFO in some cases). The classic symptom is that random DTMF is lost
when dialing.
On Mon,
Tim Sailer wrote:
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
I've heard t
Tim Sailer wrote:
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
That's just the way Asterisk's dial command works.
Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP
On Monday 09 February 2004 14:42, Bisker, Scott (7805) wrote:
> Has anyone experienced problems with dialup through asterisk. I'm
> having some varied success with dial-in and dial-out.
>
> All my analog extensions are connected to * via Adtran 750 FXS
> channelbanks using FXO_KS signalling. I ha
Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
As a redhat fan, I'd say Fedora. With that said, I've got a DL380 running
RH 8.0 (updated) that is ro
On Mon, Feb 09, 2004 at 03:51:51PM -0500, William Waites said:
> On Mon, Feb 09, 2004 at 09:31:30PM +0100, Philipp von Klitzing wrote:
> >
> > Why exactly would hylafax be a "worst case" solution only, why would you
> > tink that that the Asterisk solution is better at all?
>
> The "worst case"
> On Monday 09 February 2004 00:26, Soragan wrote:
> > > Ohh.. You better give that to me then. I'll send you my Pentium
> > > 133 w/ 16 megs of ram. It works great with the X100P.
> >
> > LOL, can your Pentium do web server, mail server with spam and
> > virus checking and ADSL router all together
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:9242(256) to H323:28967(8)
Feb 9 15:37
On Mon, 2004-02-09 at 14:42, Bisker, Scott (7805) wrote:
> Has anyone experienced problems with dialup through asterisk. I'm
> having some varied success with dial-in and dial-out.
>
> All my analog extensions are connected to * via Adtran 750 FXS
> channelbanks using FXO_KS signalling. I have a
Alright,
I have quite a problem on my hands and even the digium engineers are
stumped. First my system layout
Asterisk CVS-01/15/04-16:44:03 built by [EMAIL PROTECTED] on a i686 running Linux
X101P cards: 3
SNOM200 Phones: 5
2 outgoing lines and 1 incoming line (DSL)
Okay, so on
Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
As others have said: Whatever you're most comfortable with. Having said
that though, I'm partial t
Would the following be a doable solution:
1. An Asterisk-box on site with FXS
2. Plug Fax into FXS
3. User uses facsimile machine to call a number - Asterisk answers
4. Stores called number into variable ${FAXDESTINATION}
5. Use RcfFax of * to store fax within asterisk
6. mail stored fax together
On Mon, Feb 09, 2004 at 09:31:30PM +0100, Philipp von Klitzing wrote:
>
> Why exactly would hylafax be a "worst case" solution only, why would you
> tink that that the Asterisk solution is better at all?
The "worst case" would be the modem hairpinned into an FXS
port, not hylafax per se.
> > I
Hi!
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered SIP to be seen
> as unavailable. Am I just missing something obvious, again?
You are right, th
Has anyone experienced problems with dialup through asterisk. I'm having some varied
success with dial-in and dial-out.
All my analog extensions are connected to * via Adtran 750 FXS channelbanks using
FXO_KS signalling. I have a longdistance T-1 (e&m_w) from sprint and a local T-1 PRI
from V
Plug the Packet8 DTA310 into an X100P and go.
On Mon, 2004-02-09 at 14:06, Jim Flagg wrote:
> - Original Message -
> From: "Eric Wieling"
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 23, 2004 11:07 AM
> Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
>
>
>
Don't allow=all. Don't ever allow=all. In fact don't even think about
allow=all. Personally I would like the allow=all option REMOVED.
disallow=all
allow=ulaw
If that works then allow whatever codec you want instead of ulaw.
On Mon, 2004-02-09 at 14:01, Ryan Courtnage wrote:
> There appears t
Hi!
> What I would do i this situation is work out a fax <--> email gateway.
> Best case this could be done entirely with software on the asterisk
> box, worst case a faxmodem hairpinned into an fxs card using hylafax.
Why exactly would hylafax be a "worst case" solution only, why would you
tin
> I had two system freezes this weekend, first time. I just setup
> Musiconhold. The kernel panic referenced mpg123. I turned off
> musiconhold until I could look into it more.
Again, please post your comments *on the tracking bug* number 963, at
bugs.digium.com. Please include whether you have S
I would like to request an alternative to the mpg123-only musiconhold. I
could live with just about anything that isn't mp3. Just as an alternative
for those of us who have had nothing but problems with mpg123 and would like
to use the functionality of musiconhold.
MATT---
-Original Message--
Please include your sip.conf and extension.conf files. Hard to say what is
wrong without seeing the configuration
- Original Message -
From: "Bill Michaelson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 09, 2004 3:15 PM
Subject: [Asterisk-Users] asterisk-grandstream
On Mon, 2004-02-09 at 13:54, Mark Spencer wrote:
> > Currently in progress of trying to debug similar
> > problem on my own system. Sometimes it happened
> > during call transfers, but this last time,
> > it happened all by itself at 4:00 AM, no calls even
> > close. Complete system Freeze, Noth
I am trying to muddle my way tthrough getting something - actually
anything to work - with Asterisk. I've acquired a Grandstream phone and
I've got * on a Red Hat 9 box. I've gotten to a point where I can see
(via ethereal) that the phone REGISTER's successfully with asterisk, and
then I try
I had two system freezes this weekend, first time. I just setup Musiconhold. The
kernel panic referenced mpg123. I turned off musiconhold until I could look into
it more.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Mark Spencer
- Original Message -
From: "Eric Wieling"
To: <[EMAIL PROTECTED]>
Sent: Friday, January 23, 2004 11:07 AM
Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
> The Packet8 8x8 DTA-310 that I have ran SIP when I was using it.
Eric, did you get the DTA fully working w
There appears to be a problem with Asterisk negotiating a codec with my
x-lite clients (both mac and windows).
All codecs were enable in the clients, and sip.conf contained:
allow=all ; Allow all codecs
Using 'show sip debug' on the * console would print the following when
a c
On Mon, 9 Feb 2004 [EMAIL PROTECTED] wrote:
> How would you work the dial command to dial out using Voiceglo?
I use:
exten => _9NXX,1,Dial(SIP/1801${EXTEN:[EMAIL PROTECTED])
where 'voiceglo' is the name of the service definition in my sip.conf:
[voiceglo]
username=
secret=
...
Greg
On Monday 09 February 2004 01:20 pm, Steve Kennedy wrote:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
> Debian/Fedora Core ?
>
>
> Steve
Nah, go with good ol' DOS 3.3!
You will not have
I've looked around and found previous discussion about this, but so far I
have not seen any answers that really solve this problem.I'd like
to integrate an intercom system into Asterisk so that users could dial an
extension, the phone on the other end would emit a beep, and then the
speakerphone wo
> Currently in progress of trying to debug similar
> problem on my own system. Sometimes it happened
> during call transfers, but this last time,
> it happened all by itself at 4:00 AM, no calls even
> close. Complete system Freeze, Nothing at all
> workings, except the reset button.
>
> You set
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
> That's just the way Asterisk's dial command works.
Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP channels register a
The Cisco 7960 and 7940's have a two-port SWITCH built into the phone, at
least according to Cisco they do.
John
Paul Mahler wrote:
The phones I am familiar with have two ports combined as a hub.
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
__
That's just the way Asterisk's dial command works.
On Mon, 2004-02-09 at 13:16, Tim Sailer wrote:
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered S
have you tried this gs-102 with pppoe? verizon dsl uses pppoe. pppoe is
logically like dhcp, but using ppp for added feature like aaa :) can this
unit connect directly to a cable modem?
- Original Message -
From: "Nicolas Bougues" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monda
-do you use hyperthreading
-do you use the LD_ASSUME_KERNEL=2.4.1 b4 loading asterisk
-have you compiled zaptel with the SMP flag on
Can anybody site some real hardcore technical facts
about SMP & hyperthreading support in the RH9 kernel rpm images
I hear what i would call 'old wives tales' about
You must register with cisco in order to get ata image.
- Original Message -
From: "Anton Tinchev" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 09, 2004 8:48 AM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
> Billy Huddleston wrote:
>
> >Anyone ha
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
Tim
--
>
We've had 2 "unexplainable" system freezes.
We have SMP and Redhat 9 2.4.20-20.9. There has been no evidence
anywhere of why our system crashed.
Tan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Welter
Sent: 09 February 2004 18:39
To: [EMA
Hi!
> I have worked a little on debugging the Grandstream cfg.txt file and
> here is what i found out so far:
Have you check this?
http://www.voip-info.org/
tiki-index.php?page=Asterisk+phone+grandstream+budgetone
You are strongly encouraged to add any additional findings yourself on
the Wiki
> The phones I am familiar with have two ports combined as a hub.
>
> Paul
OK, but how would an scenario look like where You only would be "allowed"
to use one authorised MAC? Is it possible or forget it and try to get that
NAT stuff working!!
/t
>
>
> Paul Mahler
> mail:[EMAIL PROTECTED]
> pho
Did you possibly have astman running on the localhost? I found that I was getting
kernel panics while using astman on an SMP machine with dual T400P cards. Did you see
the message on the console before you reset the box? Did you possibly have a serial
console connected logging console message
Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
Which ever one you are most happy with is probably the best answer..
_
On Mon, Feb 09, 2004 at 06:20:54PM +, Steve Kennedy said:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
> Debian/Fedora Core ?
Not dumb, but you won't get a scientific answer. You wil
what you feel most confortable with.
there's a best linux distro for *.
Matteo.
Il lun, 2004-02-09 alle 19:20, Steve Kennedy ha scritto:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
>
Johannes von Drachenfels wrote:
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14
[..]
But i still can see only the 78107-0 when
On Mon, Feb 09, 2004 at 02:28:02PM +0100, Dawid Mielnik wrote:
>
> Would the t.38 transmission be properly handled by the t.38 supporting end
> points whith mediastrem passing through Asterisk ? (dont have much
> experience with t.38) Has anyone ever tried anything similar / different /
> wierder
How would you work the dial command to dial out using Voiceglo?
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Friday, February 06, 2004 8:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE:voiceglo sip config
Hi,
Aft
Did you have any active meetme sessions at the time of the freeze?
What Asterisk version are you using?
MATT---
-Original Message-
From: Jonathan Biggs [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System freeze
Cu
RH9. If it happens again I think I'll drop back to 0.7.1.
Jonathan Biggs wrote:
Currently in progress of trying to debug similar
problem on my own system. Sometimes it happened
during call transfers, but this last time,
it happened all by itself at 4:00 AM, no calls even
close. Complete syste
> The way I found out was I installed the new firmware and watched my TFTP
server :)
>
Maybe they are on the way to support APS configuration as sipphone.com have
made on www.plugndial.com (hope so)
Jens
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Asterisk-Users mailing list
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I'm having bad echo between TDM and
SIP. There's no echo between TDM-TDM though. I've seen this post from
JTodd:
; Config notes:; - in /usr/src/zaptel/Makefile, set
KFLAGS+=-DECHO_CAN_MARK2; - in /usr/src/zaptel/Makefile, set
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR ;; I compile with these tw
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
--
NetTek Ltd Phone/Fax +44-(0)20 7483 2455
SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19
_
use call files there is should a sample in the asterisk src
On Mon, 2004-02-09 at 12:21, John Chambers wrote:
> Newbie question coming up ...
>
> Is it possible to use the asterisk to initiate a call to a phone?
>
> What I'm trying to determine is ways for software to connect to a
> phone and
Currently in progress of trying to debug similar
problem on my own system. Sometimes it happened
during call transfers, but this last time,
it happened all by itself at 4:00 AM, no calls even
close. Complete system Freeze, Nothing at all
workings, except the reset button.
You setup is vastly di
The way I found out was I installed the new firmware and watched my TFTP server :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill
Sent: Monday, February 09, 2004 11:51 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firmware for
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Chambers
> Sent: Monday, February 09, 2004 12:21 PM
> To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Can asterisk make a call to a phone?
>
>
> Newbie question coming up
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