Dear Girish,
Thank you for the comments, my responses are inline.
My question:
CAN I use asterisk software as a replacement for siprg ? CAN I use DIGIUMs
x100p PCI card as a replacement for QUICKNETS CTI LINE JACK CARD ?
Asterisk is a soft PBX that has a SIP channel. It doesnt
Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream
uses the latest firmware and SIP INFO.
Iain
--On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED]
wrote:
I cannot get the Message Waiting Light (MWL) on my Grandstream phone to
turn on when I leave
Iain Stevenson wrote:
Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream
uses the latest firmware and SIP INFO.
Iain
--On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt
[EMAIL PROTECTED] wrote:
I cannot get the Message Waiting Light (MWL) on my Grandstream phone
Howdy,
The first 5 secs of each Voicemail or IVR announcement is stuttered and u
can hardly hear the sound. After that its ok.
Running TOP showed a high CPU usage on start up of the announcement as
running command X??
Is this a PC CPU/RAM issue or something else related to Asterisk
OS :
Anand,
My question:
CAN I use asterisk software as a replacement for siprg ? CAN I use
DIGIUMs
x100p PCI card as a replacement for QUICKNETS CTI LINE JACK CARD ?
Asterisk is a soft PBX that has a SIP channel. It doesnt perform all the
functionalities of a sip router. You can use digium's
Title: Mensaje
Hi
all,
For a few days we have a veryextrange
problem. We have an intranet with Budgetone and others SIP Phones.
In the
extranet We HaveBudgetone Phones. The whole system was working well
between the extranet and the intranet until a few days ago.
When
we try to speak
I have several Sipura's on our network. The
are FXS devices. So you will not be able to connect your Pot's line to
it. The do not support GSM they use ULAW/ALAW, and 729.
There are ways to configure your phone but you need
to reconfigure your wiring. I don't have the link to it but there
Show us your extensions.conf
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Talk about a stoopid question...
How do I exit the CLI of Asterisk. Typing exit (per the pdf manual and
my google results) brings up a message saying QUIT and EXIT are no
longer available, that STOP NOW is used to shutdow the pbx.
I do not want to shutdown the pbx. I just wanna get outta the CLI
How to use Standard 802.3af POE Injectors w/ Cisco 7940/60
It was recently brought up on the IRC channel that the 7940 and 60
phones from Cisco do not support the 802.3af Power Over Ethernet
standard. At first, I didn't believe it, so I grabbed a 7960 and
tried it with a standard 3COM POE
I looked at my sample config's and I cannot find an example of an extension
where you specify the codec differently for each extension. Can someone show
me a sample extension?
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 26,
In a nutshell: Can I use Asterisk to hook up an intercom at my front
gate? My wife would like to have one of those simple
speaker/microphone intercoms. People show up at our front gate, press
the doorbell, it rings in the house. We pick up a phone on my Asterisk
system and dial (example) 105
G'day David,
On Thu, 26 Feb 2004, David J Carter wrote:
Q) Is ADSL a standard? and will his router/modem work in AU?
There are a few standards for xDSL. ADSL down here uses the G.Lite
standard (maximum speed 1M5/256kbps), so as long as the modem supports
that it'll be fine technically
I'd suggest disabling g723 in grandstream or disallowing it in asterisk.
- Original Message -
From: Paul Zimm
To: [EMAIL PROTECTED]
Sent: Friday, February 27, 2004 7:11 AM
Subject: Re: [Asterisk-Users] Grandstream - firefly call translator problem
I also ran sip debug. The output is
Joel Maslak wrote:
My understanding is that the purpose of the button is to look pretty
unless you have the higher-end budget-tone (102?) where it then does 3 way
calling.
Doesn't work on my BT102 phones.
Michael
--
Michael T Farnworth
Maxima Systems Ltd (http://www.maximasystems.com)
16
I cannot get MWI working either with GS101 firmwire 1.0.4.39
My sip.conf has the mailbox number specified. voicemail.conf has mailbox
set up. I have collecting mail fine.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
smime.p7s
Description: S/MIME Cryptographic Signature
Even though it was 100, I'm also keen to hear
about large installs, what kind of experience
did you have setting it up, and what hardware for
the * server did you use?
Thanks in advance
Barry
Matthew wrote:
I've set up 75 extensions... I'm 100.
Sorry.
Sincerely,Matthew
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Nice. Thanks. I was unaware of this hardware. It looks like
The freeze-ups were due to a NetGear NIC card. Haven't had a freeze
since I removed that card.
Mike
Steve wrote:
On Monday 09 February 2004 11:45 am, Michael Welter wrote:
I have a Gigabyte K7 motherboard with an Athlon 2400+ processor.
Before the T1 install I had two T100P cards, one for
I've set up 75 extensions... I'm 100.
Sorry.
Sincerely,Matthew MarloweGear 3 Technologies,
LLC609.252.1155
x614www.gear3.com(00) Choose a
job you love, and you will/||\ never have to work a day in your
life.=/\=
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
You must have started asterisk with asterisk -c so you cannot bail out of
CLI with exit -- you are in console mode. Instead, start it without -c so it
respawns another service process and exits to shell, after that you can run
asterisk -r and bail out with exit all you please ;-).
- Original
We are connecting all the lines via ISDN-PRI to TN767 boards.
Matt
On Feb 26, 2004, at 9:14 PM, htguy wrote:
How are you connecting to the definity? Through analog/digital trunk
ports,
analog station ports or digital station (BRI) ports using an Eicon
Card?
I only have Partner and Magix
Hello,
Has anyone gotten chan_h323 and
chan_oh323 to run on the same system at the same time? Provided you change the
listening ports of course. I can get both of them to start, but whenever I try
to make a call using chan_h323 I get a segmentation fault. This doesn't happen
if I disable
You can do this, and I've posted how to do it... Although I've been
called idiotic when I said it. Amazingly enough it was working for me
so not so idiotic.
Good luck on your ventures.
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
Choose a
Transfer it into a call park
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have the following in my sip.conf entries:
callerid=Anonymous 8885551212
This still passes the number for 911, but flags the call as private. I
believe this will meet your requirements.
Stephen
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Thursday,
Ok, I Clipped this from the tek-tips forum for definity and thought it
might help you with your definity CID issue.
FYI the url I got it from is
http://www.tek-tips.com/gviewthread.cfm/lev2/9/lev3/89/pid/690/qid/742640
-Art
r3jnp1 (Programmer) Jan 22, 2004
Can you send
I've searched several times for help on configuring a front gate
intercom through Asterisk, but I haven't come across anything. If this
is a repeat post (as well it could be due to the amount of traffic this
list experiences), sorry.
In a nutshell: Can I use Asterisk to hook up an intercom at my
On Thu, Feb 26, 2004 at 12:43:27PM -0500, Steve Creel wrote:
On Thu, 26 Feb 2004, Rob Fugina wrote:
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
Olle E. Johansson wrote:
Going back to the subject, what does the grandstream really do,
SIP-wise, when you press
the transfer button?
Olle,
The following is an exact transcription of the description given in the
BT101 manual for Blind Transfers:
4.3.7 Call Transfer The user can transfer an
I've also included output for call from Zap channel to firefly that
works fine
here is output from iax2 debug
==
-- Executing Macro(SIP/mhorst-1f03, ext|IAX2/[EMAIL PROTECTED])
in new stack
-- Executing DBget(SIP/mhorst-1f03,
I am using mini-itx motherboard and I installed asterisk stable from
cvs. However below is the messages when starting asterisk by
safe_asterisk. Anyone spotted the cause of not starting.
Last login: Fri Feb 27 10:40:44 2004
[EMAIL PROTECTED] root]# safe_asterisk
[EMAIL PROTECTED] root]#
The problem turned out to be in my voicemail.conf, thanks. I had the second
section named [bell] instead of [default]. The MWI works perfectly now with
both the Grandstream and IpDialog SIP phones.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle
Hi all users,
Can build a switchboard with TDM400P + X100P?
I need a receptionist to pick up the incoming calls and transfer them to
appropriate employee.
Do I need those Nortel telephones for this or Panasonic KXTD kind of phones?
Can I use an ordinary touch-tone phones to transfer the
Key word: input.
My telephone line input is my x100p fxo card, and it is a ONE-port
card. I have no unused line input on my phone system. Therefore, I'm
hosed with these models???
Help me... I've fallen, and I can't get up...
with this unit
On Thu, Feb 26, 2004 at 11:16:19AM -0500, Jon Pounder wrote:
(at my door you can knock, ring a doorbell, or pick up the door phone -
you would be surprised how many people knock. Probably there are some
that
are scared off entirely that I don't even know about.)
It seems to me that placing
Hi, I can answer part of the Caller ID question.
Also, at least in the testing I've done, the text portion
of
the CLID string is ignored by the telco. They only look
at the number and
generate the text based on what is in their database. IE;
If I tell my
asterisk server to set my
Periodically when users delete voicemail a file
gets left behind that triggers an inaccurate message waiting light. Users
attempt to pickup/erase what they think is a legitimate message.
/var/spool/asterisk/voicemail/default/*/INBOX/msg.txt
Thanks for your help.
Rob
On Thu, 26 Feb 2004, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 12:16:06PM -0800, TC wrote:
http://www.vikingelectronics.com/products/app-notes/doorboxes.html
The W-1000, W-2000A and W-3000 doorboxes are designed
to be installed on the unused telephone line input of nearly any phone
According to someone else here, that would be idiotic.. (Altho my idea
was to put it into a call park where you can than pick the call up.) Or
write an AGI script to transfer the call back to the original person that
just transferred the call away. But once again, that must be idiotic.
If you
Dear Mark
We have a customer who would like an
Asterisk server setting up. Do you provide this service, please? I read in a
news posting that you could provide remote support?
Regards
JB
JohnBenson
Managing Director
Solutios Ltd
10
dkwok wrote:
I cannot get MWI working either with GS101 firmwire 1.0.4.39
My sip.conf has the mailbox number specified. voicemail.conf has mailbox
set up. I have collecting mail fine.
If you're running any other voicemail contexts other than default (in your
voicemail.conf), you need to
On Thu, Feb 26, 2004 at 01:20:00PM -0600, Rob Fugina wrote:
Yup, that looks like the right device to me... Looks like it'll connect
to an FXS port on your * box. Wonder what that thing costs... Looks like
you need 3 dates with a sales rep before they'll quote you a price...
Matt wrote:
Hello All,
I was wondering If you can specify which voice codec is used per
extension. I'm using sip phones that support gsm, and some H.323
Endpoints that support GSM, and a couple that don't, and with oh323
codec negotiation doesn't work properly. So I'm wondering if I can
Hey Guys,
As part of our legacy integration I am trying to get our lucent
definity to pass caller id information from internal stations to
asterisk, I have no problem getting it from lines passed in from the
telco, but the internal stations/vdns etc just don't do it. Anyone have
any
Steve Dolloff wrote:
I have the following in my sip.conf entries:
callerid=Anonymous 8885551212
This still passes the number for 911, but flags the call as private. I
believe this will meet your requirements.
Stephen
OK. I was under the impression that the PSAP got their information based on
When I try to initiate a call from my Grandstream phone (ext 8010) to my
firefly softphone (ext 8030) I get the following error messages, but I
have no problem calling from firefly ext to grandstream ext. Calling
from a Zap phone to firefly works fine also.
Feb 26 07:25:47
Had the same issue. My Mini-itx board has a VIA C3 processor and this fixed
this issue for me.
Set PROC in the main Makefile of asterisk to i586 then recompile.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dkwok
Sent:
Actually, it works fine as long as the parkpos values are numbers. If you
put in a * or #, it seems to ignore what you supply and start with 701. I
just happened to be starting with a *.
-Original Message-
From: Jim Sneeringer
To: [EMAIL PROTECTED]
Date: Wed, 25 Feb 2004 13:48:47 -0600
Stephen R. Besch wrote:
Olle E. Johansson wrote:
Going back to the subject, what does the grandstream really do,
SIP-wise, when you press
the transfer button?
4.3.7 Call Transfer The user can transfer an active call to a third
phone by using the Transfer button. The sequence is like this: The
How are you connecting to the definity? Through analog/digital trunk ports,
analog station ports or digital station (BRI) ports using an Eicon Card?
I only have Partner and Magix Systems to test on, but when I get the new
boards I ordered I'll try it out on the Magix with a few configurations. If
Some of the door phone systems are designed to share an already existing line.
For example:
http://www.sandman.com/pdf/Page21.pdf
I believe some of the Viking configurations can do this too. For example
see the diagram here:
http://www.vikingelectronics.com/products/pdf/c-2000(dbb).pdf
There are many options for remote support including Digium directly or
3rd party consultants that are on this list
On Thu, 2004-02-26 at 10:09, John Benson (Solutios Ltd) wrote:
Dear Mark
We have a customer who would like an Asterisk server setting up. Do
you provide this service,
Would anyone care to share some experience with big
installs, ie. multiple PRI's and excess of 100-200 extensions.
Thanks
Rob
On Thu, Feb 26, 2004 at 09:04:52AM -0600, Greg Kedrovsky said:
I've searched several times for help on configuring a front gate
intercom through Asterisk, but I haven't come across anything. If this
is a repeat post (as well it could be due to the amount of traffic this
list experiences),
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
On Thu, 26 Feb 2004, Greg Kedrovsky wrote:
In a nutshell: Can I use Asterisk to hook up an intercom at my front
gate? My wife would like to have one of those simple
speaker/microphone intercoms. People show up at our front gate, press
the doorbell, it rings in the house. We pick up a phone on my
I just got a Budgetone 101 phone today and after configuring it I can
make calls to any other phone on my * server. The problem is that no matter
what I do, when I dial the extension assigned to the phone it will always send
me directly to voicemail with the busy message.
I tried
On Thu, Feb 26, 2004 at 10:37:05AM -0600, Rob Fugina wrote:
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
With record:
; Record voice file to /tmp
directory
exten = 9000,1,Record(/tmp/asterisk-recording:gsm)
exten = 9000,2,Hangup
Is there a way to stop recording other than hanging up?
Thanks!
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax:
Why don't you use ï in your extensions.conf to catch any invalid dialed
number and send it back to the operator.
exten = i,1,Goto(MainMenu,1000,1)
- Original Message -
From: Jim Rosenberg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 26, 2004 11:03 AM
Subject: Re:
You could always create a rule to match any-e-thing 3 or 4 digits, that
always forwards to the receptionist
[match_all_local]
exten = _NXXX,1,Goto(receptionist|s|1)
exten = _NXX,1,Goto(receptionist|s|1)
[trunk]
include = localnumbers
include = match_all_local
include = international
include =
Hi!
Thanks for the info. Which phones support consultation transfers? The
Grandstream and IpDialog phones most certainly do not.
Can you expand a little on the IpDialog phone?
Thx, Philipp
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Paul Mahler said:
With record:
; Record voice file to /tmp directory
exten = 9000,1,Record(/tmp/asterisk-recording:gsm)
exten = 9000,2,Hangup
Is there a way to stop recording other than hanging up?
Thanks!
Press the # key.
Below is from my extensions.conf. It plays the
Hi folks,
OK... I've successfully managed to get a DTA-310 from 8x8 to take
inbound calls from the PSTN, into an AS5300, through asterisk. I can
call the number associated from out there in the world, it rings, I
can speak to the person on the other end just fine. However, I cannot
seem to
Transfer it into a call park
So you are suggesting to transfer ANY caller to put them into a call park?
That's idiotic -- The OP was asking how to get the caller back after a
mis-transfer. If you park every single call you're sidestepping the issue
entirely.
Regards,
Andrew
Hi,
I own a Siemens 3070 DECT system.
It's a simple DECT base which allows the connection
of a few DECT phones. It's a very basic PBX.
It's connected to the public network using an ISDN
bri (2B + D) plug. According to the doc, it can also be connected to a
PBX.
Is there a way to connect
Going back to the subject, what does the grandstream really do, SIP-wise, when you
press
the transfer button?
/O
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Did y ou email this directly to me? Sure I do
that
Sincerely,Matthew MarloweGear 3 Technologies,
LLC609.252.1155
x614www.gear3.com(00) Choose a
job you love, and you will/||\ never have to work a day in your
life.=/\=
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
strange, do a iax2 debug to see what codecs firefly is asking for.
- Original Message -
From: Paul Zimm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 26, 2004 11:42 PM
Subject: [Asterisk-Users] Grandstream - firefly call translator problem
When I try to initiate a
Yes asterisk works with the transfer button
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
Hi friends,
I am experiencing lot of duplicate digits especially when people dial-in
using Cellular phones.
here is my config;
PSTN(PRI)---Asterisk A---(IAX)-Asterisk B--SIP Phones,
H.323 G/W
Asterisk A switches calls to Asterisk B. Asterisk B answers the call and
That still isn't my point.
Nevermind, I give up.
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
I also ran sip debug. The output is listed below.
=
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060
SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
sip:[EMAIL
Hi!
For a few days we have a veryextrange problem. We have an intranet
with Budgetone and others SIP Phones.
In the extranet We HaveBudgetone Phones. The whole system was working
well between the extranet and the intranet until a few days ago.
What did you change a few days ago?
When
But... my wife wants an intercom. So, I'll probably hunt down one of
those nice, 200-dollar Viking jobs that's basically a telephone that
looks like an intercom. Best of both worlds. The only ouch is the
stinkin price tag.
This has been on this list b4 there are less expensive devices that do
On Thu, 26 Feb 2004, Rob Fugina wrote:
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Hi,
does anyone know hardware, which supports an S2M
(E1/PRI) via C7 (CCIT7) (instead of DSS1) and is
supported by asterisk?
Roger.
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On Thu, 26 Feb 2004, Rob Fugina wrote:
On Thu, Feb 26, 2004 at 10:19:08AM -0600, Greg Kedrovsky wrote:
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Nice. Thanks. I was
Can someone tell me how to respond to a list message? If I e-mail to the
list, does it always start a new thread?
Thanks.
Jim
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For some reason voicemails are not being played back. I can log into the
voicemail system and i get the menu. its all fine till the point asterisk is
announcing info abt the mail but when it comes to playing the mail i hear
nothing then it quickly gones on to the next mail. If anyone has
According to someone else here, that would be idiotic.. (Altho my idea was to put it
into a call park where you can than pick the call up.) Or write an AGI script to
transfer the call back to the original person that just transferred the call away. But
once again, that must be idiotic.
Title: Video Recording
Hi,
I'm trying to record a video in asterisk.
In order to do that i use the record aplication: record(/tmp/video:h263).
After i call the extension using messenger 4.7 i get a h263 file.
I was wondering if asterisk recognizing the h263 format is the only
thing it
Hi all,
i have a TDM20B in my astbox and i have configured my channels as follows:
usecallerid=yes
signalling=fxo_ks
context=tel1
group=5
callerid=101
channel = 13
callerid=102
channel = 14
But if i make a connection to the manager interface the callerid in the
events is not set:
Event:
At 00.04 25/02/04, Jean-Denis Girard wrote:
Robert Sprockeels a écrit :
Hi, good solution, I think I will do something similar ... but can you also
dial out from your home with the right MSN or only the main MSN is sent
over outbound calls ?
; Appel de la
Olle E. Johansson wrote:
Stephen R. Besch wrote:
Olle E. Johansson wrote:
Going back to the subject, what does the grandstream really do,
SIP-wise, when you press
the transfer button?
4.3.7 Call Transfer The user can transfer an active call to a third
phone by using the Transfer button. The
On Thu, 26 Feb 2004, Jeremy Jones wrote:
I _think_ my problem has to do with the Dial Plan settings on the SIP
configuration page. Anyone familiar with these things? By default, the
dial plan setting reads: 1xx|x.T.
This is my dialplan for Packet8 / 8x8:
exten =
On Wed, Feb 25, 2004 at 05:14:35PM -0500, I wrote:
So: now I've got my caller just sitting there, transferred into nowhere.
Is there a way to pick the caller up? I haven't found a way to do this.
Sorry to be a nag, but no one answered the original question. Is there a
way to pick up a stranded
On Thu, Feb 26, 2004 at 03:49:11PM -0600, James Sizemore wrote:
You could always create a rule to match any-e-thing 3 or 4 digits, that
always forwards to the receptionist
This has the same problem as a catch rule -- suggested in other posts --
for the invalid extension. I don't want to catch
Rana Dutt a écrit :
I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn
on when I leave a new voice mail message for that phone. I have specified
the correct mailbox in my sip.conf as follows:
[200]
type=friend
username=200
host=dynamic
context=dialout
callerid=200
You can only use the r option if you answer the call first
exten = 106,1,Answer
exten = 106,1,Dial(SIP/106,30,tr)
other wise remove the r
Erick Weber V. wrote:
Thank you very much
I just make the change and I'm up an running.
One more quick question, why I can not hear the ring in the phone
Has anyone heard when Digium is scheduled to release the FXO modules for the
TDM400P? How about the expanded TDM card (I believe I heard it was to be 12
ports)? Or the legendary IAXy?
Thanks,
Steven Sokol
Owner/Manager
Sokol Associates, LLC
Phone: 816.822.1807
IaxTel: 700.613.9004
Web:
Hey All,
I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940
phones.
My issue is that from what I see in chan_sip.c there is no support for the
Remote-Party-ID field in relation to withholding the calling partys number. This is a
legal requirement for many
On Thu, Feb 26, 2004 at 12:16:06PM -0800, TC wrote:
http://www.vikingelectronics.com/products/app-notes/doorboxes.html
The W-1000, W-2000A and W-3000 doorboxes are designed
to be installed on the unused telephone line input of nearly any phone
system or...
Key word: input.
My telephone line
On Thu, Feb 26, 2004 at 09:12:08AM -0800, TC wrote:
This has been on this list b4 there are less expensive devices that do the
same thing
http://www.at-fairfax.com/Intercom/DoorbellFon.htm $105 for ctrl/door box
and add another $40 box for Electrical Lock Controller if you want pop the
door
On Thu, Feb 26, 2004 at 10:28:37AM -0500, Steve Creel wrote:
As for hardware, take a look at:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Nice. Thanks. I was unaware of this hardware. It looks like something
similar to the Viking W-1000 would work perfectly. Press
When I speak of the dial plan here, I'm referring to a portion on the
DTA-310 web pages, not my * dial plan. I've seen a couple posts about
setups like this:
* w/tdm card -- dta-310 -- packet8 network -- pstn
I'm not using the packet8 service, just the gear. Like this:
dta-310 -- * -- as5300
On Thu, Feb 26, 2004 at 09:04:52AM -0600, Greg Kedrovsky said:
I've searched several times for help on configuring a front gate
intercom through Asterisk, but I haven't come across anything. If this
is a repeat post (as well it could be due to the amount of traffic this
list experiences),
On Thu, Feb 26, 2004 at 11:10:03AM -0600, Rob Fugina wrote:
Am I getting things backwards, or is the W-1000 what he was talking
about as an FXO device. I'm having trouble finding an FXS version on
Viking's site at the moment...
Looks like the FXS device they offer is the one that
Hi John-
If you monitor this list, you'll find that many of the people on it can help
your client set up an asterisk server. Also, there is a list of consultants
on the Wiki, some are European-based. Try this page:
http://www.voip-info.org/wiki-Asterisk+consultants
My own clients are mostly
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