Colin Anderson wrote:
If five people in the office all need to use their phones at the same
time, would I need five VoIP lines, or would I only need one VoIP line?
Am I over-thinking this?
You would need 1 broadband connection, and technically, you would need only
1 ACCOUNT (I think that's the wor
Andrew Kohlsmith wrote:
On December 6, 2004 10:12 pm, Michael Giagnocavo wrote:
Except the providers who offer "unlimited" -- in that case, they want you
to use as little as possible, so they can make their money.
They're the ones that are on the way to bankruptcy.
EXACTLY ;)
Aint no free lunch, m
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with
zaptel, etc. IAX as the * <-> Channel bank protocol.
Just an idea...
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Paul Rodan wrote:
We have a customer that handles the billing for a rather large company.
Anyway, they have their phone system through us, Cisco 79xx phones with
Asterisk and such. They want us to build them an IVR system that can
interact with their billing system through XML and read back informa
news.gmane.org wrote:
nik martin wrote:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling
with zaptel, etc. IAX as the * <-> Channel bank protocol.
Just an idea...
Allied Telesyn VoIP Access Devic
Walt Reed wrote:
On Wed, Dec 08, 2004 at 08:43:10PM -0600, nik martin said:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with
zaptel, etc. IAX as the * <-> Channel bank protocol.
Yes. Search th
Jean-Michel Hiver wrote:
What leds are lit?
Looking with the orange bit facing you, the network led on the left
(network) is permanently lit. The led on the right blinks once every 7
seconds or so. There is also the network plug's led which is lit. That's
all.
What kind of phone is connecte
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kobus
Wolvaardt
Sent: Wednesday, December 22, 2004 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about VOIP phones
hi,
> Is it sure, that you n
find many
places in the USA that sell it but nowhere in the UK.
Also can anyone recommend some good VoIP suppliers in the UK?
Thanks,
Martin Mitchell
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Is it possible to listen on more than one port within a single instance
of *? I have an engineer in Iraq that we need voice comms with, but the
gov't limits traffic to ports 80,443, 25, and 110. Can I set up IAX to
listen on port 80 AND the regular IAX port?
Or will I have to set up some weir
>
> Does anyone know of Phone that supports G.723 on H.323.
>
Innovaphone tiptel 200 for example.
http://www.innovaphone.com/webneu2/products/en_IP200.asp
One of the nicest phones I've seen so far, h.323 only though.
Bye, Martin
__
e in extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possible?
Thanks,
Nik Martin
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Ahh, the {EXTEN:1} must serve to skip the 1st character of the extension,
stripping the 9 off
That's what I needed, thanks!
-Original Message-
From: Austin M. Brower [mailto:[EMAIL PROTECTED] On Behalf Of
Austin M. Brower
Sent: Tuesday, April 13, 2004 9:11 AM
To: Nik Martin
Subjec
it looks like some other usb module tries to get loaded and that's what
causing it.
try to insmod the zaptel & tor2 & run ztcfg -vv instead.
or rmmod all the uhci modules...
regards
Martin
On Fri, 16 Apr 2004, Jim Gottlieb wrote:
> We use dual Athlon machines with up to three
notransfer might be still a [global] only keyword for IAX2.
regards
Martin
On Tue, 20 Apr 2004, Hans-Henrik Andresen wrote:
> Hi,
>
> Another one.
>
> I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get
> this in my logfile
>
> Attempting native br
Hi list,
surely this has been posted before but the archives don't offer a
'search' functionality and I need an answer really soon on this
subject... so, my apologies.
Which ports (range) must be open on a firewall, either TCP and/or UDP,
for Asterisk to work correctl
s this the normal behaviour?
Thanks,
Martin
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[res_parking]
'710' => 1. ParkedCall(710)
[res_parking]
'711' => 1. ParkedCall(711)
[res_parking]
'712' => 1. ParkedCall(712)
[res_parking]
'713' => 1. ParkedCall(713)
[res_parking]
'714' =>
A tip to avoid much Head-On-Desk confusion: The MWI light will only light
up on cisco phones ( and all other MWI equipped phones) if the phone is in
SIP context 'default' using the form:
Mailbox=123
Otherwise, you must use:
[EMAIL PROTECTED]
I went around and around with this for 5 days until I
need to add a hangup after the VoiceMailMain I also think
> exten => o will work in that case too ... not sure from
> VoiceMailMain but you could try it.
>
> bkw
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PRO
Out of context, this isn't much information. Is your network connection OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Iain Steve
Does anyone have any recomendations for a free
Windows softphone, SIP or IAX that supports the following features:
* Message Waiting Indicator
* Consultative Transfers
* Speed Dials
This drove me crazy for a while too. See my post, here:
http://lists.digium.com/pipermail/asterisk-users/2004-May/046959.html
Your sip.conf must include the context of the mailbox for the user if it
isn't in [default] context
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EM
Post your zapata.conf and zaptel.conf
Nik
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
> Sent: Thursday, May 20, 2004 9:45 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] x100p card + dailing out
>
>
> I think I have it configu
What address is that? Is it a phone (or address of a PC with a softphone?)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Dolloff
> Sent: Thursday, May 20, 2004 10:41 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Mystery SIP chan
What is the best way to upgrade a production asterisk box? make upgrade? I
don't want my configs messed with, and need the process to go as smooth as
possible. I fetched and built a new kernel last night, but haven't rebooted
into it. I'll do that tonight, and then want to quickly upgrade to th
Is it normal for asterisk to have to be recompiled when you upgrade your
kernel? I fetched a new kernel yesterday, taking me from 2.4.25 to 2.4.26
I built yesterday, and rebooted this afternoon. After upgrading, none of the
asterisk modules would load. I assume they are dependent on the kernel th
M
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk upgrade on production box
>
>
> Yes you will need to recompile zaptel also. This is comonly
> talked about on the mailing list.
>
> bkw
>
> - Original Message -
> From: "Nik Mar
I upgraded to the latest HEAD version of asterisk, and all IAX calls started
sounding choppy. It was suggested on the IRC channel that I go back to
asterisk -stable to determine if that fixes it. Is downgrading as simple as
upgrading? Because now, -stable builds fine, but I get an error on the
a
undefined symbol: ast_get_txt
> May 25 23:28:42 WARNING[16384]: loader.c:408 load_modules: Loading
> module app_txtcidname.so failed!
>
> jo
>
> Nik Martin wrote:
>
> >I upgraded to the latest HEAD version of asterisk, and all IAX calls
> >started sounding c
Sorry, that's illegal. You have to purchase the support options via Cisco
that entitle you to software upgrades. It's $8.50 per phone through most
retailers, but it takes 6-8 weeks for cisco to issue you a password.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECT
rules somewhere? It
has to.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin
> Sent: Tuesday, May 25, 2004 2:53 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Downgrading Asterisk
>
>
> I upgraded to t
access
for you, after we have a short phone conversation.
Nik Martin
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rich Adamson
> Sent: Friday, May 28, 2004 7:59 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] D
The disconnect between HEAD and stable is what concerns me. The fact that a
fix was put into Stable for the choppy audio on Cisco <-*->IAX that I
couldn't find in HEAD, and that didn't work when fetching and rebuilding
HEAD is what concerns me. If it exists in stable (and works in stable), but
do
As a sidenote, your site doesn't work in Mozilla Firefox.
>
> --
> Vice President of N2Net, a New Age Consulting Service,
> Inc. Company
> http://www.n2net.net Where everything clicks into place!
> KP-216-121-ST
>
>
__
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Olle E. Johansson
> Sent: Friday, May 28, 2004 9:38 AM
> To: [EMAIL PROTECTED]
> Cc: Asterisk-a-users-list
> Subject: [Asterisk-Users] Re: [Asterisk-Dev] Time to lock down v1.1?
>
>
> On the othe
Steven refers to the safe_asterisk script.
I call it from my own rc.asterisk that also loads all the proper modules:
rc.asterisk:
#!/bin/sh
# load the modules first
modprobe wct1xxp
modprobe wcfxs
/sbin/ztcfg
#now start asterisk, via the script
/usr/sbin/safe_asterisk
# end of rc.asterisk
O
Tony Hoyle wrote:
No idea what you mean by PBX class telephone but if anyone at our
company spent $500 on a phone they'd probably be fired (unless it was
the boss).
Our desktop phones were done as a package deal from the building owner
(who also runs the existing PBX) for almost nothing.
Al
Oh and you need a fine digium card to interface with the channel bank.
Nik
Nik Martin wrote:
Tony Hoyle wrote:
No idea what you mean by PBX class telephone but if anyone at our
company spent $500 on a phone they'd probably be fired (unless it was
the boss).
Our desktop phones were done
> > BTW: And are you sure people wouldn't like to have
> voicemail? You'll
> > need
> > to make them want that... ;-> I guess you can even argue
> that voicemail
> > increases productivity.
>
> Since we share phones (at least the developers/non customer facing
> people) voicemail wouldn't wo
All those numbers kinda negate the whole purpose of 3 digit nationally
standardized numbers, huh?
>
> emergency call numbers - without engagement ;-)
> Sorry for my bad translation ...
>
> 112 european emergency call
> 120 car breakdown service
> 122 fire department
> 123 car breakdown service
>
Are your phones SIP? I have a configuration like:
Sip.conf:
[nmartin]
Mailbox=105
Callerid = "Nik Martin" <105> <<<< THIS MUST BE DEFINED
Etc.=etc.
Yada=yada
Extensions.conf:
Exten => 105,1,dial(SIP/nmartin,20,tT)
Exten => 600,1,Wait(1)
Exten => 600,2
The main window needs to be sizeable. I have a dual monitor workstation
that this would run on, and only want it on one monitor, while other apps
reside on the other. Also, what is your licensing? This might be a nice
addition to the Open Source community for * windows developers.
Nik
>
PROTO=UDP SPT=5060 DPT=62975 LEN=456
--
Why is this happening? We got no relationship with the DST IP address
and external access is not allowed.
Any ideas?
Martin
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Has anyone ever thought configuring asterisk on a pair of pc's to act as
remote broadcast terminals for the broadcast radio industry? Seems like
a stripped down asterisk on a laptop with a PCMCIA ISDN modem connecting
to another asterisk instance on a PC at a radio station would work
nicely.
There are commercial providers online that build ready-to-go asterisk
servers and hardware:
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934066432.htm
They should be able to build a turnkey solution for you. There are also
consultants on this board that will probably assist you i
t; Why? Just use shoutcast/icecast for that.
>
> Bkw
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Nik Martin
> > Sent: Friday, June 04, 2004 7:49 AM
> > To: [EMAIL PROTECTED]
>
oadcasting,
etc.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Nik Martin
> Sent: Friday, June 04, 2004 9:40 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] (possibly) new use for asterisk
>
>
> Does shoutcast
Title: Message
H,
Google
is your friend:
http://www.google.com/search?q=SIP+phones+asterisk&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8
The
second result brings you to a page that's all about your
question.
It
also links to a HUGE resource list:
http://www.voip-in
Tony Hoyle wrote:
That's no help.. read all of them. The best I can find out is the $8
price on the wiki is bogus and should be removed as it's misleading.
The cheapest smartnet is CON-SNT-PKG1 at $75 per year. That's almost
ten times what I expected I'd be paying.
Not true. I just bought th
I'd do three separate 2 U Rackmounts, loaded with ASUS MB's with SATA raid
controllers, onboard nic and video
3.0 gig 800mHz FSB p4's
Slackware Linux, latest 2.4 kernel
1 gig ram
2 120 gig sata drives each server, in raid 1
1 t100p each
1 TA750 connected to each box
YOU NEED REDUNDANCY WITH THAT
>
> Would this mean three separate voicemail systems? Why not diskless
> servers with /var on an nsf mount from a file server?
>
> --
Good point, a fourth server with the sata raid subsystem would offer a much
more efficient and administratable system.
__
Hi all,
can Asterisk be used as a videoconference server or the like when
enabling 'videosupport=yes' ? if so, how do I use it? is there any
recommended SIP/Video-client for both Windows and Linux?
Thanks,
Martin
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Asterisk-Users mailing l
The WIKI is your friend:
http://www.voip-info.org/wiki-Asterisk+record+calls
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of hank smith
> Sent: Monday, June 07, 2004 5:10 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Network Sniffing
Has anyone managed to get Early-Dial working with
the grandstream phones?
On my older phones running firmware 1.0.3.X it
works fine, but it doesnt work on the newer versions..
3750 so information covering
this model would be nice too...
Do you know if any of the PBX listed on the link above are similar
somehow to the Siemens I mention in terms of integration with Asterisk?
Answers much appreciated.
Martin
___
Asterisk-Users
Is there a clever way to camp on an extension in asterisk? What I need is a
way to answer my extension (not just a ringing ZAP channel) from any other
phone. If I'm in another office and hear my phone ringing, I want to be
able to quickly pick it up from that extension. The list revealed the
pic
Jason Williams wrote:
I recommend a PRI E1 link into the Hicom from *
Thanks for your reply and excuse the following silly question: and then,
what? :)
Martin
Jason
At 16:05 08/06/2004 +0200, you wrote:
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with
Down here.
>
> It seems to be down, I even tried dialing for
> example 1-800-555-TELL. I tried yesterday
> and again today.. Just get dead air.
>
> Stephen Rosebush
>
> Mark Musone wrote:
>
> >Does anyone know if the 1-800 iaxtel gateway is down?
> >I've been trying to use it all day today and
ePyron Felix Deierlein wrote:
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with VoIP from PC to PC, we'd like that our
people abroad could dial company internal extensions through Asterisk
using a SIP client. On a second approach, the
Need a good document for the Manager API before a GUI can be written!!!;)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Pablo Endres
> Sent: Wednesday, June 09, 2004 11:35 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] NetworkWorld
ecret=**
pickupgroup=1-4
;The phone attempting the *8
[nmartin]
type=friend
host=dynamic
insecure=no
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.100
dtmfmode=inband
mailbox=105
context=Outgoing
callerid="Nik Martin" <105>
username=nmartin
secret=**
pickupgroup=
Didn't know everyone was down on it. It's just not a very used feature in
my office environment. What's needed is a true camp-on. That's used lots
at everywhere I've ever worked, and asterisk is missing it. It has an
anemic call pickup that doesn't do much for us. (or even work at the moment)
Look in the asterisk source directory for a file called sample.call
Read it and it'll give you all thed details
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Simon
> Sent: Thursday, June 10, 2004 10:28 AM
> To: Asterisk-Users
> Subject: [Asterisk-
It doesn't seem possible to get that call
transferred TO my extension, using the example context that accompanies the
Redirect sample on the WIKI:
[transfer]
exten => _.,1,Dial(Zap/g1/${EXTEN})
It doesn't look dooable.
Any ideas?
Nik Martin
___
If I have an IAX client (Firefly softphone in this example), and the client
is not registered at the moment because they are not connected to the
network and someone dial that extension, they get the user's "I'm on the
phone at the moment" message vs. the "I'm unavailable" message. Is this by
desi
>
> I need clarification as to DID in T1 connection.
>
> T1 provides 24 channels for voice/data. Do it assign each channel to
> particular DID. Or you can have unlimited DID to share the 24
> channel as
> an example. ie. Outgoing/incoming traffic is not bound to particular
> channel. Whateve
Title: Message
Let's
say you have company 1 with a did of 256-704-2000, and company 2 with did of
256-704-3000
In
your dialplan:
extensions.conf:
[default context]
exten
=>s,1,Answer
exten=> 2567042000,2,Goto(company1,s,1)
exten=> 2567043000,2,Goto(company2,s,1)
[company1]
coma
Title: Message
You
can capture user input with the background application
You'd
probably have to write your own app that validates the account number, returning
0 if the account is invalid, -1 otherwise, etc.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PRO
Easy now, that just showed up today, the old "NEW" website never worked,
and rates WERE hard to find. Don't defend it, it makes it worse than it
really is.
Nik
Brian K. West wrote:
Look at this from nufone.net:
We provide IAX and SIP termination.
US48 termination for 2. cents (USD) per minu
Michael Bielicki wrote:
>> - CTI support (dialing from within Outlook using hardware VoIP
>> phones)
> there is a project for that which sems to work although we havem't
> tested it yet
I'm using asttapi https://sourceforge.net/projects/asttapi/ and it works
fine.
> Well, yes, you're sort-of dreaming.
>
> The trick is not designing the hardware or the software - anyone with
> $100k (or much, much less) and the right engineers can get something
> working to the point where it is ready to be produced.
>
> You will hit the wall with:
>- finding reliable s
Kevin P. Fleming wrote:
> I have an asterisk server up and running, using Firefly in IAX mode
> works great, even with Firefly behind a NAT (as expected, since IAX
> works really well with NAT).
I have the same scenario, but after about 4 hours, the Firefly phones can
still make calls, but asteris
Of course, big brother wants his say in the matter.
http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead
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> Notice there need not be ANY telco POTS lines.
>
>
> I wonder if there is a group discussion of this type of functionality.
>
> Would the LINE OUT/IN from Asterisk to analog MIXER console be PC
> Sound cards or something more discrete like a form of telco line
> cards?
> We do not need t
I would guess that you have Echo squelch enabled in capi.conf. Either
disable this or use app_capiNoES (In the latest capi build) before
forwarding a call to voicemail. Beware, app_capiNoES does not check for
channel type so only use it on a CAPI channel.
Cheers
Martin
Andrew P Cook wrote:
> I am looking to install a new PBX into a small business. We have 18
> internal extensions, and 6 phone lines. I have been looking at
> Asterisk as a possible solution and would like to hear from people
> already using it. Digium recommended I post to this list for
> respon
Jeremy Kenney wrote:
> I am new to asterisk I just downloaded it I setup some extensions I
> can't seem to get them to ring I can get my ata 186 to register but
> having problems with getting the phones to ring when I dial an
> extention
>
> Extentions.conf
>
> [dstech4]
> exten=>104,1,Answer
Adam Goryachev wrote:
On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote:
So, if someone could brief me on the GPL issue, and (perhaps someone
else) offer a distribution point, it's free for the asking, VB sources
and all.
Stephen R. Besch
Alright, I've waited a long time before offering thi
I have had very bad experiences with IAXYs so far.. I have pulled them
and will be attempting a refund shortly. Bad audio, overheating and
shutting down until allowed to cool, etc. make it unusable in a business
environment.
That said, is there a low-mid priced solution for a remote office to
Ronald Wiplinger wrote:
Nabeel Jafferali wrote:
I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623] ; IAXy
type=friend
host=dynamic
accountcode=aaabbb
disallow=all
allow=ulaw
secr
7;s only one PRI provider where we are currently)
Any help would be appreciate :-)
Thanks
Martin Roy
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will start with channel 4 and not
pick a random one between the 3 channels as I'm pretty sure if I put in
my dial plan a group having channel 2, 3 and 4 it might do the opposite
and start with channel 2 then if it's busy switch to 3 and then 4
instead of 4 then 3 then
I have been getting the following message in Asterisk and it shuts Asterisk
down, needing a reboot.
"Power alarm on Module 2"
I have
(1) TDM400P with (2) FXS & (2) FXO cards
(1) X100P card
Any ideas?
Thanks
Martin
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Yes, The card is working fine most of the time. It just gets this message on
occasion and then Asterisk shuts down. I debating putting surge suppressors
on the PSTN lines. Could this be caused but a voltage issue from the Telco?
Martin
-Original Message-
From: [EMAIL PROTECTED
I'm using a server with dual AMD opteron processors with a TDM04B
without any problem. The server is running Fedora Core 3 AMD 64bits.
Hope this answer your question...
Martin Roy
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SIP for a start.
Then I have to install another Asterisk server at another location and connect
the 2 together. I'll probably install the same server at the second location
with a T1/E1 card instead of the TDM04B cards.
Martin Roy
From: "Robert Augustyn" <[EMAIL PROTECTED]>
S
n 7.3. But I had a hard
time upgrading from 2.1 to 3.3 as the 3.3 image as more then 8
characters and firmware 2.1 support only 8:3 (8 characters plus 3
characters for the extension)
If you need any help let me know.
Martin Roy
___
Asterisk-Users ma
7.1 there)
http://ns.goodgrief.com/voice-comm/ (there's 2.x,3.x and 4.x there
but the 4 didn't seem to work for me)
As weird as it might seem you can also find some on eDonkey and Kazaa if
you are willing to wait a day or so to download them...
Hope this help
Martin Roy
Martin Roy wrote:
Get a
Well the best solution would be to create a VPN between your network and
the one of your customer but that's only possible if you have a VPN
router on both side. Otherwise I don't see much solution then the one
you already consider doing.
Martin
From: Michael Welter <[EMAIL PROT
I'm wondering why are you using SCCP and not SIP as most of us that use Cisco
7960 phones?
Martin
From: "Nenad Radosavljevic" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Problem with chan_sccp and cisco 7960
Hi !
On Cisco 7960 (with or witho
t I would still need to provide the SIP firmware and all the other
files open on the internet...
In the Cisco phone I don't even see a way to change the default TFTP
port but even that solution doesn't seem good enough...
Anyone having
?
Thanks
Martin
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Sorry forget my post I forgot to add it in zaptel.conf... now it's
working fine... That's what happen when you want to do things a little
too fast hehe ;-)
Martin
Martin Roy wrote:
I already have one Digium TDM04B installed in my server working fine.
I just received 2 more so I did
t one. But for incoming calls how can I
setup asterisk to answer on the first 10 lines with one message and on
line 11 and 12 with another one?
If I put the s,1, Answer thing it will answer all 12 lines with the same
message...
I'm sure it's easy but I just don't know how to
So if I understand well this should do the trick : (be aware that
context first and second include all my extensions that I haven't
included in this and in my SIP phones use context firstinternal and
secondinternal)
zapata.conf :
context=firstincoming
switchtype=national
signalling=fxs_ks
echot
difference to answer PSTN calls.
Thanks
Martin
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y
extensions start with the number "2" if I add a choice that when someone
press 2 it dial some SIP extensions will it take the 231 if I type it or
it will say it as a 2 and forget the other digits?
Thanks
Martin
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I tried to upgrade to 1.0.5 for libpri and all
asterisk stuff to the last version. But now, my spans stays in RED Alarm. All
seems ok, no error. Just no more links up
I use a TE405P.
Any idea ?
I think it’s something from the old release
that make conflict.
So I tried to get back to
List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Zaptel down after upgrade.
On Mon, 2005-02-07 at 20:16 +0100, Régis MARTIN wrote:
> I tried to upgrade to 1.0.5 for libpri and all asterisk stuff to the
> last version. But now, my spans stays in RED Alarm. All seems ok, no
>
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