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
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
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
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
JVSen
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
___
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[EMAIL PROTECTED]
http:/
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 a
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: [E
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]
http
Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for
that.
The Linux bit is all free, and only a couple of PCB work to disenable the
protection.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Albertson
Sent: 03 February 2004 1
Matteo,
try: -
[incoming]
include => default ;default location for internal phones
exten => s,1,Answer
exten => s,2,Wait 10
exten => s,3,Dial(SIP/100)
exten => s,4,Hangup
Make sure that the context of incoming is defined in zapata.conf for pstn
calls.
Dave
-Original Message-
From:
Have a look at http://www.plugndial.com/aps_sample.html
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 09 February 2004 17:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firmware for Grandstream Phones -
Supports CFG by MAC ad
If you add
include => context-of-normal-extensions
at the beginning of you MENU section then this should work.
[mainmenu]
;
;"main menu" context with submenu
;
exten => s,1,Answer
include => default
;exten => s,2,SayDigits(${CALLERID})
exten => s,3,Background(hello_and_thank_you)
exten => s,4,Wa
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]
Subje
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] softphone
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 cr
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 fo
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: [Ast
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 rid of the slash and number on
| the register
| line,
| | unless your asterisk ex
bject: Re: [Asterisk-Users] Anybody 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 wrot
x27;t 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
|
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, updates,
Mark,
Zaptel is where it is told to go.
I have mine set to incoming, and a context of incoming in my
extensions.conf.
My configs are here http://www.codepipe.com/id25.htm without any UID's or
PWD's.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Beh
Brian,
You need to put an include => default in your incoming context.
some samples here http://www.codepipe.com/id25.htm
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Mulligan
Sent: 02 March 2004 17:58
To: [EMAIL PROTECTED]
Subject:
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
T
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 PROT
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 ne
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 X100
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 PROTE
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.
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
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
-
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
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
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
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
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
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
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
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
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 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
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
] 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
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
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
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 don’t, or doe
: [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
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
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
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
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
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
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.
Als
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 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
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 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
_
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,
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
[
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
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
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
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
; 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
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
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
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 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 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
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 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
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
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
-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 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
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 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
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
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
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
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
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
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
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 u
Just
one question, Why do you want users sent to the Demo at
Digium?
take a
look at: - http://www.codepipe.com/id25.htm I
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
Da
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.
Hi all,
I may need to connect to a system with E&M 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]
http://lists.digium.com/mailman/listinfo/
rch 2004 15:35, David J Carter wrote:
>>I may need to connect to a system with E&M 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.
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 entering
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
T
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