Hi,
Have you got port 4569 open in your NAT/Firewall?
I take it that your extension ranges on the servers are 5000 and 6000 range.
The configs look OK, same as mine, and mine works fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
David,
Try something like this:-
zapata.conf
context=me
signalling=fxs_ks
channel = 1
;
context=her
signalling=fxs_ks
channel = 2
;
context=fax
signalling=fxs_ks
channel = 3
;
context=meandher
signalling=fxs_ks
channel = 4
extensions.conf
[me]
exten = s,1,Dial(SIP/0001,30,t)
exten =
The only Caller ID phone I can get to work on the TDM card is one with
belcore caller ID, the UK callerid phones do not work here.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
Sent: 27 January 2005 17:47
To: Asterisk Users Mailing
Hi,
Try something like these, works for me.
extensions.conf
[general]
;
static=yes
;
writeprotect=no
;
[globals]
;
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;
#include globals.conf ;This includes your conf file with your fqdn's
listed.
Hi all,
I have just been reading an article on the asterisk-doc site about ISDN
X-Over cables.
The article mentioned the converting of an NT1 to make this possible, has
anybody got the information required to modify a BT NT1?
Or any information on the BT NT1.
Thanks in advance.
Regards
Dave
Stefan, Peter,
Thanks for the replies guys.
I have looked at the web page and will work on it over the weekend.
My next step will be to find out hoe the CO lines connect, but that's
another project.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
One thing I do on remote sites is set up a soft phone so I can call myself,
this proves out the link and quality before anything else. DIAX id good for
this as you can connect to multiple sites, also good to see if you have
problems before anyone else calls you to say there is a problem.
It also
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To:
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Luis,
Am I right in thinking that the MVP400 is the non SIP MultiTech box.
The SIP version I think is the MVP410.
You could load the H323 stack on the box and use H323 to connect to
Asterisk.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
How do you want Switch to appear to Asterisk.
1. As an extension. Then use an FXS connection to a CO line input.
2. As a CO line. Then use an FXO connection to an Extension output.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of [EMAIL
In your [mainmenu] use the include = context_for_internal_numbers, or at
least the ones you want peaple to call.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philip
Siegrist
Sent: 11 February 2005 15:58
To: Asterisk Users Mailing List -
It means for some reason you lost your CO line for 10 Seconds.
Either someone pulled the plug out by mistake or the Exchange line went away
for 10 seconds.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anton Krall
Sent: 23 February 2005 09:35
To:
All,
I have downloaded and installed openh323 as per the documentation.
When the machine now reboots safe_asterisk just keeps restarting.
If I start another session and just load asterisk -vvvgc asterisk loads.
If I enter noload chan_h323.so in the modules.conf then safe_asterisk will
kick
Guy,
I
think what Lyle meant was to put a wait as in dial -- wait ---
number.
Therefore the line is seized and then after a wait the number is
dialled.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Guy C.
GuckenbergerSent: 27
*CLI sip show peers
Name/usernameHostDyn Nat ACL Mask Port
Status
176polycom 192.168.0.176 255.255.255.255 5060
Unmonitored
175polycom 192.168.0.175 255.255.255.255 5060
Unmonitored
Added to sip.conf:
Nigel,
I have
bugetone phones working with 2, 3, 4 + extension numbers.
Check
you config's, or post them here and lets see if we can find the
problem.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Nigel
BurgessSent: 04 March 2005
I have used the Draytek 2600V router in a few locations where only 1 or 2
phones are required.
The router has 2 FXS ports and can be used locally to an * box or via the
VPN to a remote * box.
The VPN built into the routers just works, and I have 1 user who has had 3
VPN circuits up and running
Nigel,
Should really be on the biz list for this, but Telappliant sells Digium
hardware.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nigel
Taylor
Sent: 05 March 2005 21:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Darrell,
You could try talking to Telappliant, (in London like yourselves), I use
them for one of my connections and have found them very good.
ISDN is the best way to go if you are looking for your own PSTN connections
and to cut down on hardware in the machine I would be looking at an ISDN-30
I have had the same problem.
Just uploaded 1.0.4.40 and all seems OK again.
Dave
[EMAIL PROTECTED]
SIPPhone: - 1 747 669 1957
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton
Sent: 15 January 2004 21:18
To: Asterisk List
Subject:
Hans,
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik
Andresen
Sent: 19 January 2004 08:43
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
This is the URL I got the config file from, http://www.plugndial.com/ it's
on a link from the SipPhone URL.
I just modified the text for my phone.
There is a bit more info on there, and there is a MAC address on the top
line of the file.
Still just playing with this myself so don't know all the
Hi All,
In my extensions.conf I have : -
exten = _6XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _6XXX,2,Playback(remote_unavail)
exten = _6XXX,3,Hangup
;
exten = _7XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _7XXX,2,Playback(remote_unavail)
exten = _7XXX,3,Hangup
;
exten =
Hi all,
Since my upgrade to CVS dated 14-01-2004 I am unable to call or receive
calls through my ZAP channel.
When calling out I get the following message: -
WARNING [155667]:app_dial.c:527 dial_exec: Unable to create channel of type
ZAP
In zaptel.conf
fxsks=1
loadzone=uk
defaultzone=uk
In
Deepak,
I am
using X100P on a telewest service with no problems at all.
Contact me off list and I can send you a copy of my
configs.
[EMAIL PROTECTED]
Regards
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar
JVSent: 29
Hi all,
I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.
Anyone done this or know how it's done.
regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Thanks John,
I think it is not that simple. I am not using a phone but a Cisco ATA.
The scenario: -
User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension
The Multitech MVP100 used to connect to my old analogue switch which was set
to
James
I would have to change several other units over from proprietary to h323
that are already in the loop.
I added mine to the loop so they could call for support.
I have started to play with h323 on the * but not got my head round it yet.
Regards
Dave
-Original Message-
From:
Thanks John,
Found it.
The Multitech's are part of a legacy system used by a new customer of mine.
I just latched onto it for ease of communications, it's been in for some
years now.
Regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for
that.
The Linux bit is all free, and only a couple of PCB work to disenable the
protection.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Albertson
Sent: 03 February 2004
Matteo,
try: -
[incoming]
include = default ;default location for internal phones
exten = s,1,Answer
exten = s,2,Wait 10
exten = s,3,Dial(SIP/100)
exten = s,4,Hangup
Make sure that the context of incoming is defined in zapata.conf for pstn
calls.
Dave
-Original Message-
From:
Have a look at http://www.plugndial.com/aps_sample.html
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 09 February 2004 17:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firmware for Grandstream Phones -
Supports CFG by MAC
If you add
include = context-of-normal-extensions
at the beginning of you MENU section then this should work.
[mainmenu]
;
;main menu context with submenu
;
exten = s,1,Answer
include = default
;exten = s,2,SayDigits(${CALLERID})
exten = s,3,Background(hello_and_thank_you)
exten = s,4,Wait,t,2
I had this problem with an old 16bit Sound Blaster Card.
Threw the card away and put in a cheap ?3.50 PCI card.
Works a dream now.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Boardman
Sent: 15 February 2004 23:20
To: [EMAIL PROTECTED]
I ordered a WiSIP from them on Friday last, and had confirmation yesterday
the it was in Transit from the US to The UK.
E-Mail them they are very good at responding.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jonathan
Moore
Sent: 18 February 2004
I noticed you had collerid not callerid in the conf file.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: 18 February 2004 19:57
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
I had this after my last CVS update.
A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman
Sent: 24 February 2004 19:17
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to
Hi,
Sorry for the of topic question, but where else do you get so many telco
guys in one place.
I have a customer who is moving to Australia and was on ADSL here in the UK.
Q) Is ADSL a standard? and will his router/modem work in AU?
I have told him a tentative yes but would page the oracles
Hi,
I would be tempted to get rid of the slash and number on the register line,
unless your asterisk extension is 02115800.
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: [EMAIL PROTECTED]
Subject:
) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get
place calls (via
sipgate.de) I don't think it is a firewall matter...
Birk
David J Carter wrote:
| Hi,
|
| Are you behind a NAT/Firewall?
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
| Sent: 28 February 2004 11:04
| To: [EMAIL
Hi Johnathan,
I wouldn't mind a copy of the firmware if you could send it.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jonathan
Moore
Sent: 28 February 2004 19:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] wisip firmware,
No need to string them together.
Just put them in the MP3 directory and it will play them one by one, taht's
all i have done.
My largest MP3 plays for 20 minutes.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dean Collins
Sent: 04 March 2004 07:05
Tony,
Have a look here http://www.codepipe.com/id25.htm these are my working
examples.
I have 6 GS phones. The GS set-up's are from extersion 8002 onwards in
sip.conf.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony
Mountifield
Sent: 06
Simon,
Caller ID does not work in the UK, well not on my BT or Telewest line's.
Have a look at my sample configs http://www.codepipe.com/id25.htm , I am
also in the UK and these work for me.
Give me a call if ya want to chat about it.
Regards
Dave
-Original Message-
From: [EMAIL
Hi All,
I have now got my '*' server up and running quite good.
As stated in earlier posts I am no Linux guru, so a bit of hand holding
required.
First Subject.
I would now like to add h323 boxes to the '*' server, I have looked through
the wiki and followed the instructions about what I
hi,
Try
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of randulo
Sent: 12 March 2004 14:54
To: asterisk list
Subject: [Asterisk-Users] X100P and TDM400 questions
I have the dev kit installed and the
Chris,
May be a bad card, or more likely Microfilter, I have had mine on the same
line as the ADSL for 3 months now and no problems.
As for UK CLI I will be glad when I can get CLI from either BT or Telewest.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
My aim is that, i want to connect my PC (where i
installed the asterisk) to another PC in my network
for voice chating. For this purpose, what are the
steps to
be done? which are the files to be modified. I would
like to make use of the existing Hardware (sound card,
network card etc), i am not
Just
one question, Why do you want users sent to the Demo at
Digium?
take a
look at: - http://www.codepipe.com/id25.htmI
have some sample files there.
If you
want to contact me off list [EMAIL PROTECTED] the we
will not tie the list up with 8000 posts for every reply.
Regards
Dave
John Lawrence wrote
Hi all,
Does anyone know the procedure for adding a serial output to a cheap caller
display unit. If I can find a way of doing this then I'm sure there will be
away for linux to take the CallerID info, write it to a file, * to open
that
file an read the number from it.
TIA
Hi all,
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
Regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
:35, David J Carter wrote:
I may need to connect to a system with EM connectivity.
Am I right in assuming a T1 card and Channel Bank will give me this
connectivity?
Perhaps we ought to make sure we're talking about the same thing.
Mr. Carter: are you talking about EM signalling on digital trunks
Wim,
If ya don't need callerid then add the patch at
http://www.nodomain.org/asterisk to zaptel and asterisk directories.
I did this for UK callerid and the phone now rings on the first ring of the
CO.
Bit of a bodge but it works.
Dave
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I had the same problem until I changed iax.conf to not have a callerid=
field in it for the context you are using.
All I have now is.
[guest]
type=user
context=default
I have several servers all talk to each other, and get caller/extension ID
from them all.
Dave
-Original
You
haven't made my mistake and forgotten about case sensitivity in Linux have
you.
I had
the same problem when I called mine Mainmenu and put mainmenu in the
dialplan.
Regards
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tim
I have it working with the X100P no problems, on both BT and Telewest lines.
Anybody got it working on the TDM400P yet?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan
Natesan
Sent: 17 June 2004 19:59
To: [EMAIL PROTECTED]
Subject: Re:
Don't you need a 'modprobe wcfxs' also?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Lewis
Sent: 18 June 2004 14:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problems with X100P
All,
I'm having trouble getting the X100P working.
Hi,
Does anyone know of a provider/terminator of Belfast, Ireland telephone
numbers?
Thanks in advance
Regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Title: Leterhead
Hi
I am a newbie and just set up my first Asterisk box.
I have got 2 x Grandstream 101s working as extensions and am now
looking to get to the outside world.
Q.) Can you use a voice/fax modem as an FXO interface?
If yes, then how would I configure it.
Hi all,
Has anyone had chance to connect one of these units to the *, if so how u do
it?
Cheers
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
: David
J Carter
To: [EMAIL PROTECTED]
Sent: Friday, October 10,
2003 2:05 PM
Subject:
[Asterisk-Users] X100P Config
Hiya all,
I have just received my X100P telco card and I dont seem to be able to
talk to it.
I am a bit of a numpty on Linux being from the Windows (wash my
to do it:
http://www.digium.com/index.php?menu=faq#Configuration_7
- Original Message -
From: David J Carter
To: [EMAIL PROTECTED]
Sent: Friday, October 10, 2003 5:29 PM
Subject: RE: [Asterisk-Users] X100P Config
Hi,
I can see the card with a cat /proc/pci.
I don't seem to have
Hi again,
When I run modprobe zaptel I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB while this kernel version is
2.4.20-4GB-athlon. And fails.
When I run modprobe wcfxo I get the message that the zaptel.o was
compiled for kernel version
Hi All.
When I run modprobe zaptel I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB while this kernel version is
2.4.20-4GB-athlon. And fails.
When I run modprobe wcfxo I get the message that the zaptel.o was
compiled for kernel version 2.4.20-4GB
-
From: David J Carter
To: [EMAIL PROTECTED]
Sent: Friday, October 10, 2003 5:29 PM
Subject: RE: [Asterisk-Users] X100P Config
Hi,
I can see the card with a cat /proc/pci.
I don't seem to have a zaptel.conf file in the etc directory.
Dave
-Original Message-
From: [EMAIL PROTECTED
Thanks Rich,
I am re-installing the base SuSE Linux system again and will try to install
everything without doing any updates. I can't remember any updates being
done, but these automated installs for numpties like me could do anything
and I wouldn't know.
I will let you know how it goes.
the numeric part the of the module and not the
extra stuff)
David J Carter wrote:
Thanks Rich,I am re-installing the base SuSE Linux system again and will try to installeverything without doing any updates. I can't remember any updates beingdone, but these automated installs for numpties like me
Hi,
Anyone know if there is a problem with the [EMAIL PROTECTED] ?
I am trying to get the zaptel asterisk downloads and keep being told that
connection is refused.
Regards
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
2003 11:25, David J Carter wrote:
Hi,
Anyone know if there is a problem with the [EMAIL PROTECTED] ?
I am trying to get the zaptel asterisk downloads and keep being
told that connection is refused.
Perhaps because you need to use the host: cvs.digium.com, not
digium.com?
-Tilghman
Hi all,
Is there any way to get * to start when linux boots?
I am running Red Hat 8.0, but a remote site I am testing IAX with has power
problems and the server there keeps re-booting, would be nice if everything
started up again automatically.
I noticed this in the list the other day,
I
Cheers,
Do I add the safe_asterisk to the rc.local file?
You may tell I am new to Linux.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 18 October 2003 10:40
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Start
David J
I have put ./var/sbin/safe_asterisk in the rc.local file but it still
doesn't start.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 18 October 2003 11:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Start
David J Carter wrote
] Auto Start
David J Carter wrote:
I have put ./var/sbin/safe_asterisk in the rc.local file but it still
doesn't start.
Have you got the zaptel drivers loading at startup?
This can either be done by using modprobe commands in the rc.local or by
using the init script that comes with the zaptel
Start
David J Carter wrote:
I have put ./var/sbin/safe_asterisk in the rc.local file but it still
doesn't start.
Have you got the zaptel drivers loading at startup?
This can either be done by using modprobe commands in the rc.local or by
using the init script that comes with the zaptel source
Thanks all for the replies.
I now * starting when the machine reboots without any user intervention
required.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: 19 October 2003 03:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Title: Need to partner with someone in Hampstead London on a deal
The info
below was passed to me when looking for Digium products in the UK.
TelAppliant VoIP
Solutions (London)
Tan Aksoy
Voice: (44) 0845 004 4040
(local rate)
E-mail: [EMAIL PROTECTED]
WWW: www.telappliant.com
Title: Leterhead
Hi,
Is it possible or has anyone done it.
Can Asterisk be connected (registered) with SipPhone?
I have got:
register = 17476691936:[EMAIL PROTECTED]/7001
This is set up in my extensions.conf.
Does this look as if it should work, cos it dont, or does
To: [EMAIL PROTECTED]
Cc: David J Carter
Subject: Re: [Asterisk-Users] Asterisk to SipPhone
--- David J Carter [EMAIL PROTECTED] wrote:
Hi,
Is it possible or has anyone done it.
Can Asterisk be connected (registered) with SipPhone?
I have got:
register = 17476691936:[EMAIL PROTECTED]/7001
Hi,
I have just set up IAXTEL connectivity and I get a similar response.
I have tried to call 1800 and the * says that a connection to
IAXTEL is made but I get no ringing or anything from the remote end.
Does anyone have a 1700XXX number I can call, or can somebody call mine,
Hi,
Thanks all for help.
Working on most 1700XXX numbers now in and out, but still no go on the
18X numbers, just tried the HP sales number for a test.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: 23 October 2003
Hi
What is your Config like to connect to sipphone?
I have two sipphone numbers and I would like to talk to them from my *
server.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton
Sent: 24 October 2003 11:06
To: Asterisk List
Subject: Re:
Why not
just the Grandstream 100, 101 102 ?
Grand as
in Grandiose, Great etc.
Stream as
in that is what we are doing with the data.
Dave
-Original
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Michael Koehler
Sent: 24 October 2003 13:06
To:
Subject: RE: [Asterisk-Users] Anyone using sipcall.co.uk ? Now sipphone
On Fri, 2003-10-24 at 15:16, David J Carter wrote:
Hi
What is your Config like to connect to sipphone?
I have two sipphone numbers and I would like to talk to them from my *
server.
register = n:[EMAIL PROTECTED
Phillip,
exten = _9NX,1,StripMSD,1
Exten = _NX,1,Dial(SIP/[EMAIL PROTECTED])
Exten = _NX,2,Congestion
Should work
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Phillip Jackson
Sent: 26 October 2003 23:35
To: [EMAIL PROTECTED]
Subject:
Hi All,
Is it possible to show which line a call has come in on in *.
My scenario is 8 incoming lines, 6 lines are trunked to one number and the
other 2 are individual lines.
I would like to pass the trunked lines to one set of extensions, and the
other lines to two other set of extensions.
Title: Leterhead
Hi All,
I have a
FWD number and wish to connect it to Asterisk to receive my FWD calls.
How I do?
Is it a
register in sip.conf or iax.conf?
Regards
Dave
Registered Office: - 23 First Street, Low
Moor, Bradford, West Yorkshire,
Hi Dan,
Just downloaded 0.9.1. Works fine on test set up internally.
I get my WAN IP dynamically and have used DynDNS.org for updating a URL for
the home network. Could the registration look for this rather than a fixed
IP address?
Regards, and keep up the good work for us non techies to use.
Hi,
Try: -
exten=t,103,hangup
or
exten=s,103,hangup
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of C M
Sent: 04 November 2003 09:37
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk does not hang up
hi,
i am trying to do to autoattendant.
Pete,
I am also in the UK and I have added an include in my extensions.conf for
the file listed bellow.
exten = _15X,1,Dial,${TRUNK}/BYEXTENSION
exten = _147X,1,Dial,${TRUNK}/BYEXTENSION
exten = _NX,1,Dial,${TRUNK}/BYEXTENSION
exten = _01.,1,Dial,${TRUNK}/BYEXTENSION
exten =
I thought the standard for the UK was 11 Digits in length, (save some old
0845, 0800, 0870 numbers), but most of these are transported to normal 11
digit numbers.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Dent
Sent: 08 December 2004
I use GS 101 102, have a look at my configs at
http://www.codepipe.com/id25.htm .
Hope they help.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen R.
Besch
Sent: 23 March 2004 20:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk
Hi Gavin,
Works OK with my 128-Bit WAP.
Remove the Space or put in an underscore and try again.
Regards
Dave
-Original Message-
Gavin Adams wrote: -
Received my Pulver WiSIP phone a couple days ago. Has anyone successfully
gotten the phone to work with 128-bit WEP? I've tried
What does your extensions.conf look like?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of pesb
Sent: 29 March 2004 18:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones
-This is my 'sip.conf' file:
Try this small extensions.conf
Don't think I have missed owt.
My config files are here, you just need to add your own extension numbers.
http://www.codepipe.com/id25.htm
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of pesb
Sent: 29 March 2004 19:26
I have my RX at 4.0 ant TX at 8.0,
I get slight echo for the first 5-6 seconds then all OK.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton
Sent: 22 April 2004 17:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] inbound calls
Just tried Pulver but it's in a password protected area.
Any idea of the other places?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Williams
Sent: 25 April 2004 10:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] smallest phone
There
Mark J Elkins wrote
Um - Digium wants you to buy their hardware - but there is a CLID
issue.. would it not make more financial sense to insert a dumb ISDN
card (or two), and upgrade your PSTN to ISDN??? Would this not assist
Digium in making sure CLID worked in the UK???
Isn't this a bit like
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