Did you change the Makefile to set the processor to i586??? The Via C3
(up to the 900Mhz model) identifies itself as a i686, but misses an
instruction..
dkwok wrote:
I am using mini-itx motherboard and I installed asterisk stable from
cvs. However below is the messages when starting asterisk b
[EMAIL PROTECTED]
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Hi
I have installed the develop version of * and want to switch to the stable
version.
Can I just DL and reinstall the stable version or do I need to uninstall the
develop version ??
If so how do I do that ???
Regards
Jan Larsen
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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 placin
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 make X exte
Hi,
Can anyone tell me if SPA-2000 sipura, talk GSM and bypass from a normal PBX and Asterisk to a analog phone ?
I want to use SPA-2000 Adapter in my office with both my * and the old PBX that has in it.
Can sippura byppass the calls from both to my analog phone ?
So i can receive calls fr
Here's what I'm trying to do.
I have 2 zap channels (x100p). One is the house line, and the other is
the business line. I have call forwarding on busy setup on BOTH lines,
to call a distinctive ring number on each other line. This way, no
matter which line is busy, calls roll over to the other.
B
Hi!
> 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: Newchannel
> Channel: Zap/13-1
> State: Rsrvd
> Callerid:
> Uniqueid: 1077819120.3
My experience with AGI is
Jim Rosenberg wrote:
The Grandstream BudgeTone 101 phone has a Transfer button. This appears to
be a "blind" transfer: once you've dialed the extension to which you want
to transfer, the phone tries to do this and then "dumps you out".
You also get similar trouble if people press the transfer butto
Low, Adam wrote:
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 requirem
On Thu, 2004-02-26 at 19:43, Greg Kedrovsky wrote:
> On Thu, Feb 26, 2004 at 09:12:08AM -0800, TC wrote:
>
> I have these hooked to a fxo port
>
> FXO? I've been thinking fxs. ?? I have a 4-port fxs card, and one port
> goes to the gate. Am I (still?) confused about fxo/fxs??
Probably not. Term
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
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
> 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 experienc
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 <--> * <--
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
>
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 "telephon
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 countrie
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:h
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
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
dtmfmode
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 cat
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, 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 => _91[2-7]XXNXX
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
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 mais
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:
Ev
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 n
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.
Since
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 encountre
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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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.
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/pro
> 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 tha
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"?
>
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"
;tag=099422b3d98a1e89
To:
Contact:
C
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]
[mailt
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
coll
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]
[mailto:[EMA
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 initi
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 Beh
Going back to the subject, what does the grandstream really do, SIP-wise, when you
press
the transfer button?
/O
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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 conne
> 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 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 fi
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
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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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
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:
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: 877.4
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:
> > > http://www.vikingelectronics.com/products/doorentry/prod
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 sear
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 o
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
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), s
Would anyone care to share some experience with big
installs, ie. multiple PRI's and excess of 100-200 extensions.
Thanks
Rob
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,
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
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
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: Th
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
Hi there,
maybe kapjod or some else is interested in this? The Asterisk box runs an
"AVM Fritz! PCI" BRI card.
Philipp
(gdb) bt
#0 0x418ba070 in pipe_msg (PLCI=257, CMSG=0x80e3d88) at chan_capi.c:1369
#1 0x418bc2fb in capi_handle_msg (CMSG=0x80e3d88) at chan_capi.c:2182
#2 0x418bc3a6 in do_m
John,
I am based in Belfast and would possibly be available to provide something
for you.
Feel free to reply off list with some more details and I'll see what I can
do.
Regards,
Steve
On Thu, Feb 26, 2004 at 03:09:23PM -, John Benson (Solutios Ltd) wrote:
> Dear Mark
>
>
>
> We have a c
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
> Sen
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 WARNING[-1242334288]
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 o
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 experience
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 c
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...
http://www.gl
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 specif
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
John Benson
Managing Director
Solutio
> 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
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
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
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 call
> 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
>
> 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 http://www.doorbellfon.com/produc
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 incom
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
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]# /usr/sbi
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"
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 ac
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:
> >> > http://
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 f
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 me
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: Thurs
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] O
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
Hey all,
I'm a new Asterisk user; just testing it out right now using kphone until
my Sipura and X100P arrive. I'd like to set up outgoing calls to roll over
from my Vonage line (connected via the X100P, and possibly a second line
with a Vonage Softphone) to an iConnectHere line, but really need o
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
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 Syste
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 ;-).
- Orig
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 O
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 th
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 lo
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 Marlow
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
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 Woodbou
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