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 => _
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
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] config
t: Re: [Asterisk-Users] Re Grandstream 1.0.4.38
David J Carter wrote:
>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
>
>
>
Where do you get the latest
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: [Asterisk-Users
John,
Try these files.
They work for me.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Coll
Sent: 02 January 2004 23:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start
Hi,
I have my GS set to in-audio for DTMF and as bellow for my sip.conf: -
[7001] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=7001
dtmfmode=inband
callgroup=1
pickupgroup=1
disallow=all
allow=ulaw
allow=alaw
allow=gsm
I am using 1.0.4.26 and all is
Don't say that.
Does that mean that from now on we will get a voice asking if we really want
to do that at every button press?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: 30 December 2003 18:27
To: [EMAIL PROTECTED]
Subje
Michael,
A reply I received from Grandstream.
Depending on your firmware version. Firmware family 1.0.4.x is not
interchangeable with 1.0.3.x and therefore cannot downgrade back. What is
the current firmware version and what version do you want to roll back to?
Regards,
Richard Huang
Grandstr
My phone's booted up and registered OK but a strange thing noticed on the
tftp uploads.
bootloader.bin
bt.bin
voc.bin
html.bin
vp.bin
ht.bin
The first phone uploaded the first four bin files.
The second phone uploaded the first five bin files.
Neither phone uploaded the ht.bin file.
Both phones a
Title: Leterhead
Mine does
that as a message indicator when mail is in the mailbox.
You get a
flashing display and a stuttered dial tone for the first few seconds.
Dave
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of bam
Sent: 24 December
Hi Tan,
Can you supply us with 1.0.4.26 firmware?
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: 24 December 2003 12:53
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
For the pri
Hi,
In rc.local I added the line /etc/rc.d/run-asterisk
I then created a small script of 2 lines called run-asterisk
#!/bin/sh
/usr/sbin/asterisk
do a chmod 755 on the file and reboot.
The Asterisk server then starts at every reboot.
Regards
Dave
-Original Message-
From: [EMAIL
Hi Rich,
For what it's worth her is an example of my IVR.
Hope it helps.
[mainmenu]
;
;"main menu" context with submenu
;
exten => s,1,Answer
include => default ; Main dialplan
;exten => s,2,SayDigits(${CALLERID})
exten => s,3,Background(hello_and_thank_you)
exten => s,4,Wait,t,2
exten => s,5,Go
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of vocalvoip
Sent: 20 December 2003 16:14
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iconnect 480 unavailable msgs
Hi guys
i signed up to iconnect a few hours ago to try do some cool stuff. but im
having a
Hi
Just a quick question on CVS.
If I want to download a CVS from say the 3rd Dec what command for CVS
checkout should I use?
Thanks in advance.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher
Sent: 17 December 2003 21:25
To: [EMAIL
Hi
I have applied the patch, I can register a Grandstream 100 from another
internet connection but I get no audio and a timeout line drop after 5
seconds.
If I call my SipPhone number 17476691936 I hear my welcome message and again
the line times out and drops after 5 seconds.
I notice that the co
Hi all
Disregard my last post I replied to the wrong e-mail, I should have replied
to an off list e-mail.
That will teach me not to put my glasses on.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson
Sent: 12 December 2003 08:01
To: [
Hi
Have you got the context set-up in the sip.conf to say which extension
context to use for incoming calls fro FWD & Iconnect.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson
Sent: 12 December 2003 08:01
To: [EMAIL PROTECTED]
Subject
Hi,
I have chan_sip.c version 1.259 do I still need the patch.
I can now get calls from sipphone.com but they drop after 5 seconds.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
Sent: 01 December 2003 18:39
To: [EMAIL PROTECTED]
Hi Nicolas,
Thanks for the file.
I would appear to have some of the file missing that the BT-100 is looking
for.
Ala,cfg.txt
sipp.bin
ring.bin
After the tftp update the program is still showing 1.0.3.81.
Any thoughts.
Regards
Dave
-Original Message-
From: [EMAIL
Hi
I wouldn't mind the 1.0.4.17 firmware.
Dave
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Glenn Dalgliesh
Sent: 05 December 2003 17:13
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone & DHCP & Ge
Hi Miklos,
I have the same as Walker.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walker Haddock
Sent: 24 November 2003 18:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iP
Hi
Didn't know there was a light under the message button, thought it just
flashed the lcd display and gave a stuttered dial tone.
This is how my mailboxes are setup in voicemail.conf
[default]
=> ,Reception Mailbox
7001 => 7001,Office SIP Phone
7002 => 7002,Lounge SIP Phone
first the
; Sub-Menu's
On Mon, 2003-11-10 at 15:28, David J Carter wrote:
> [insurance]
> exten => s,1,Background(insurance_thanks)
> exten => s,2,MusicOnHold(default)
> exten => s,3,Background(sorry_for_delay)
> exten => s,4,Goto(s,2)
> exten => s,5,Hangup
>
> if I
Hi all,
I am trying to get a Menu system to work, and having probs with the internal
extensions from the prompts.
Below is the extensions.conf section.
[mainmenu]
;
;"main menu" context with submenu
;
include => default
exten => s,1,Answer
exten => s,2,Background(hello)
exten => s,3,Background(t
Hi,
Thanks for info,
Didn't know the mails were sent as HTML, will check the email settings.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Sent: 08 November 2003 02:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Streaming MOH
It
Title: Leterhead
Hi All,
I keep asking things as they come into my head.
Is there any way to grab an audio stream and pipe it out as the MOH?
I am a helper at a local Charity Hospital Radio Station and thought it
would be nice to pipe the studio output to waiting callers.
Dave
Thanks all,
Will try to get it up and running this weekend.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos
Sent: 07 November 2003 10:46
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 Gateway
Alternatively, you may use aste
Hi all,
Anyone know of a small H323 gateway that I can run on the * box or a cheap
PC under Linux or Windoze.
I have a Multitech MV110 FXO box and would like to get it talking to *.
Any help appreciated.
Regards
Dave
___
Asterisk-Users mailing list
[
Title: Leterhead
Hi All,
This is my third attempt to get this question through.
Does anyone know of a small H323 gateway product for Linux(RH8) or Windoze.
I have a Multitech MVP120 (FXO) unit with proprietary / H323 software,
and would like to try it with *,
Thanks in antic
Hi All,
Sorry if this appears again, the first posting has not shown in 5 hours.
Is there any small H323 gateway software about for Linux (RH8) or Windoze.
Got a Multitech MVP120 FXO unit and would like to try it with *.
Thanks in Anticipation.
Dave
_
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.
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.
D
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 Yorksh
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.
Als
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:
List
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
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
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: [A
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 17
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,
170081
: [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:
>
> registe
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 don’t, or doe
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
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] A
sers] 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 scr
] 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 c
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
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
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 sugges
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
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]
http://lists.digium
only 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 num
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.
Cheers
ration_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 a zaptel.conf file in the etc directory.
Dave
-Origi
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
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 2.4.20
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 h
-
From: 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 don’t seem to be able to
talk to it.
I am a bit of a numpty on Linux being from the Windows
Title: Leterhead
Hiya
all,
I have just received my X100P telco card and I don’t
seem to be able to talk to it.
I am a bit of a numpty on Linux being from the Windows
(wash my mouth with soap and water) background, so any help would be appreciated.
I have checked under YaST2 a
Hi all,
Has anyone had chance to connect one of these units to the *, if so how u do
it?
Cheers
Dave
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Title: Leterhead
Hi
I am a newbie and just set up my first Asterisk box.
I have got 2 x Grandstream 101’s 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.
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