Matt Riddell wrote:
Hmmm...I've had 2 problem with my NuFone service in the year or more
I've used them. Each time I've treated them professionally when
reporting the issue and received the same treatment in return. The
issues were also resolved promptly.
:)
I've had no problems and hence no
Matt Schulte wrote:
you and everyone else :-)
From: Daiku [mailto:[EMAIL PROTECTED]
But i AM looking for info on another IAX capable device - like the
IAXy, but more user
friendly, as it were...
http://www.gumstix.com
There's a grass roots IAX based phone starting up using these awesome
Linux
) by
dialing 2XXX. Site two can call extensions at the master by dialing
1XXX etc.
The relatively new switch dialplan command seems like it will assist in
accomplishing this, but does anyone have a simple IAX config and
dialplan that will help me understand how this all works?
Regards,
NIk Martin
Jean-Francois Theroux wrote:
Hello,
At the office we have a Lucent PBX, which has 3 lines coming from
the CO. 2 are used for phones, 1 for fax. In the office, we have 16
phones. All those are connected in the PBX. We do not have an automated
system nor voicemail system for now. But this is
cards, but a T-1 (E-1) card may be all you
need to communicate with a legacy PBX from Asterisk. Someone with
Partner experience will certainly know more than me.
Nik Martin wrote:
Jean-Francois Theroux wrote:
Hello,
At the office we have a Lucent PBX, which has 3 lines coming from
the CO
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
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
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
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
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 Device
http
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 the list
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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[EMAIL
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
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, my
Carlos Clemares wrote:
Hi everyone,
I'm using the IAXy boxes and i'm having some trouble when I use it with
the ADPCM codec.
The IAXy only does ULAW
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-- Called 5899 at 192.168.0.5
-- Call accepted by 192.168.0.5 (format ULAW)
Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected
call to 192.168.0.5, format 0x4 incompatible with our capability
0xff03.
Hm, I'm not an expert on iaxY but it looks like that the
Erik Espinoza wrote:
Never Got that far. The device wont get an ip from dhcp for me to
provision it. . .
I use Win2k DHCP and my iaxy's grab a dhcp address fine. Do you have
ethereal (network sniffer software) on your network? If so, fire it up
and look for DHCP adresses from the iaxy's MAC
Fletcher Bonds wrote:
Hello all
As of 5pm PST today (7/13), I pulled Asterisk down off of
cvs.digium.com:/usr/cvsroot and tried to compile it on Linux ES 2.1
Actually, I pulled down zaptel, libri asterisk and compiled them in
that order as per my install guide.
When I try to compile
Gabriel Millerd wrote:
Is there a magic 'fan card' that has a power out that people are
using?
This may work for you.
http://www.thermaltake.com/products/subzero/subzero4g.htm
you lost me, its a processor cooling device. it doesnt provide
any power that could be used for a digium
Alessio Focardi wrote:
Hi,
it may be off topic, in case excuse me.
I need to get a USA phone number, possibly a Florida one.
I would like to use my actual sip phone to connect, also I would love
montly flat rates on calls.
Have you got a provider to suggest that will allow me to
CHS wrote:
is there a well-written, easy to follow, voicemail setup guide for
asterisk?
for now I don't care about understanding HOW voicemail works, I would
just like to see a quick-start guide, similar to the VOIP quickstart
guide on the voip-info.org website.
I'm about to go get an
Philipp von Klitzing wrote:
Hi!
Does anyone have a current, stripped linux distro which has only
asterisk and net drivers?
Look here: http://www.voip-info.org/wiki-Asterisk+installation+tips
and you'll find a link to the Asterisk Live! CD-ROM.
If you have a moment I guess the list
Ing. Angel Gomez wrote:
Thank's to all.
- The card came WITHOUT ANY documentation, it was not buy directly
from digium, they did not have any in stock.
- I usually go thru all the messages of this user list, maybe I
overlook at one with the same question.
- The
Eugen Cristea wrote:
Hi,
I would like to set two separate asterisks to talk to
each other.
Any suggestions?
I'm a baby asterisk fan, only started to play two
weeks ago, first managed to use kphone with asterisk
and a X100P card that is up and running as well.
Thanks,
Eugen
prepare to get flamed
Bisker, Scott (7805) wrote:
Depending on your familiarity with linux, the learning curve could be
steep and prove frustrating considering everything else you'll be
dealing with (new network infrastructure, new computers, new servers,
new telco/data circuits). Less expensive components does
You replied to a message with the subject of:
Re: Do people actually answer questions here?
And then changed the subject and started typing. This has wreaked havoc on
everybody's threaded readers, and made your question impossible to reply to.
You need to start a new message in your mail app and
Rich Adamson wrote:
From: Nik Martin [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FXO impedance matching
Date: Wed, 23 Jun 2004 11:02:00 -0500
To: [EMAIL PROTECTED]
Michael Welter wrote:
Jason A. Pattie wrote:
Robert Hajime Lanning wrote:
Echo
Jim Gottlieb wrote:
Hi all. I've been trying to build some new systems, and no matter
what I do, if I load the zaptel and tor2 drivers, the system panics
within an hour, even with no traffic.
A typical Call Trace from the panic message looks like:
wait_on_irq, [kernel] 0xde
Bonzo Armstrong wrote:
On Wed, Jun 23, 2004 at 08:24:03PM -0700, Bonzo Armstrong wrote:
On Mon, Jun 21, 2004 at 03:04:52PM -0500, Nik Martin wrote:
Try this if possible. Connect the channel bank to * via the 400' cable, but
in the same room as the * box, with all the cable coiled on the floor
GIBERT Frédéric wrote:
Hello Adam,
I'm interested by this solution, but can you please give me more info
because I don't know how to generate calls with asterisk and the
spool directory. How don't know wich files do I need to use.
Thanks.
Fred
Look in your ./asterisk directory, you'll
Michael Welter wrote:
Jason A. Pattie wrote:
Robert Hajime Lanning wrote:
Echo echo ech ech ec ec e e . .
:)
quote who=[EMAIL PROTECTED]
What's the importance of the impedance matching in a FXO interface
?
My experience is with excessive buzz and hum on the line. When I
Carlos Medina wrote:
Hi, i have a call center which receives many calls at day. Those
calls are stored in a directory in my asterisk server as WAV files.
The problem is that each call is divided in 2 files: an IN.WAV file
and OUT.WAV file. The OUT.WAV file is what im speaking to other
person,
Do you have access to a T-1 analyzer? You more than likely have a 'dirty'
T-1 line that is out of spec based on the length of the run.
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bonzo Armstrong
Sent: Monday, June 21, 2004 5:43 AM
To: [EMAIL
On Mon, Jun 21, 2004 at 12:17:38PM -0500, Nik Martin wrote:
Do you have access to a T-1 analyzer? You more than likely have a
'dirty' T-1 line that is out of spec based on the length of the run.
Sadly, none that I'm aware of, but I'll ask around. I could probably find a
decent scope
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
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
responses.
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
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 the
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
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 asterisk
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
design?
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. Whatever is
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]
comapny 1's
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
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
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:
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
___
Asterisk-Users
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
=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=1-4
callgroup=1
The SIP debug:
pbxMobile*CLI
-- Starting simple switch on 'Zap/1-1'
pbxMobile
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)
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
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 asterisk
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
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.
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 Calls
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
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
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]
Subject: [Asterisk-Users] (possibly) new use for asterisk
Has anyone ever thought
,
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 run across isdn?
-Original Message
Title: Message
H,
Google
is your friend:
http://www.google.com/search?q=SIP+phones+asterisksourceid=mozilla-searchstart=0start=0ie=utf-8oe=utf-8
The
second result brings you to a page that's all about your
question.
It
also links to a HUGE resource list:
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 work too well
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
128
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,(VoiceMailMain(${CALLERIDNUM})
Exten = 600,3
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
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
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.
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
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] Downgrading Asterisk
The code
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
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 other hand,
? 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 the latest HEAD version of asterisk, and all
IAX calls
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
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
: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 choppy. It was suggested on the IRC
channel that I go
back to asterisk -stable to determine if that fixes
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
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
: [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 Martin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 5:17 PM
Subject: RE
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 configured
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 channels
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]
)
[res_parking]
'720' = 1. ParkedCall(720)
[res_parking]
Nik Martin
Lead Software Engineer
Radiance Technologies
[EMAIL PROTECTED]
W 251.445.0045 x105
C 251.455.4665
F 251.445.0046
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[EMAIL PROTECTED]
http
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
VoiceMailMain but you could try it.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Tuesday, May 18, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoiceMailMain dumps user back into my
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
)
in front of it prior to sending the request to Voice Pulse. Is this
possible?
Thanks,
Nik Martin
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http
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
Subject: Re
91 matches
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