I'm at a loss (except) to switch Telcos, but MCI says they don't offer
disconnect supervision service on their telephone service.
I have fxsks for the FXO ports using Adit 600 channelbank with digium T1
card.
I am currently running Asterisk Stable 1.0 version.
I have busydetect=yes, but if I
Make sure you have something along the lines in your Zapata.conf file as
well.
[EMAIL PROTECTED]
callerid=Tim Thompson311
channel = 21
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan Goscomb
Sent: Thursday, May 05, 2005 11:30 AM
You will need to make sure that Transfer option in the Dialplan
i.e.
exten = 301,1,Macro(stdexten,301,${NATE})|Ttr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Goodyear
Sent: Wednesday, April 20, 2005 9:26 AM
To:
Check your firmware on the Adtran 600.
It probably needs to be updated.
Tim.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Marcin izo
Sent: Thursday, January 06, 2005 10:08 AM
To: Asterisk-Users@lists.digium.com
Subject:
Then replace ${CALLERIDNUM} with your
extension/voicemailbox and it will let you in.
No security, but sounds like you dont
want it.
Tim
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Ross Kevlin
Sent: Friday, December 17, 2004
1:22 PM
To:
[EMAIL
})
exten = s,2,Dial(${ARG2},20,rt)
exten = s,3,Goto(s-${DIALSTATUS},1)
or am i mixed up...?
On Wed, 1 Dec 2004 14:35:28 -0700, Tim Thompson [EMAIL PROTECTED]
wrote:
I have Aastra pt480e phones and would like to present the caller the CID
info about the extension being called.
I have
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Transfer
Asterisk supports blind (unattended) transfer (on SIP, MGCP and H.323) by
pressing # if Asterisk is in the media path, i.e. the Dial() statement has a
t or T in it, or if canreinvite has been set to no.
Such a transfer sets
I have Aastra pt480e phones and would like to present the caller the CID
info about the extension being called.
I have tried the following with no avail:
exten = 311,1,Macro(stdexten,311,${TIM})
exten = s,1,SetCIDName(${ARG2})
exten = s,2,Dial(${ARG2},20,rt)
exten = s,3,Goto(s-${DIALSTATUS},1)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of George Burt
Sent: Friday, November 19, 2004 2:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Basic Questions
I have installed the PCI kit from Digium and have the
You might look at installing FXS adapters
at the remote sites which would take a call from the C.O. and then pass it to
the Asterisk system at the Main site. Then you could either use the SIP phones
or IAXy adapters at the remote sites.
This would in essence terminate all the
lines for
exten = 411,1,Directory(demo|default)
exten = 411,2,Hangup
where demo is the context the voicemail boxes (/etc/asterisk/voicemail.conf)
are in and default is the context the extensions are located in from
(/etc/asterisk/extenstions.conf)
tim.
-Original Message-
From: [EMAIL
This is not an Asterisk Problem.
It is a SIP problem and most likely related to the different firewalls the
SIP phones are located behind.
You will need to configure the firewalls, use real addresses for the SIP
phones, or use IAX clients for phones behind firewalls.
Tim.
-Original
I'm here in Denver Colorado and have a pretty good cable tester that can
tell you what speed the cable will do.
Contact me off the list and I'll do what I can to help.
Tim.
Tthompson at sustain.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Centrex is a type of line and I do not believe there is a compatible card
for *.
FXS isn't going to cut it. Centrex is a digital type line.
Tim.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: Monday, November 01, 2004 1:35 PM
To:
Has anyone figured out a way to dial by last name?
I.e.
Caller hits 9 for company directory
Spell the persons Last Name. When done, press
the pound button
Caller enters 84.6#
Tim Thompson extension 300 If this is
correct, press the pound key
David Thompson extension 405
I've been pretty satisfied with the Aastra PT480.
There are some other people that say they don't like them, but I think
the $110-$120 ea. Works great for our office and the people I install
for.
Take it for what you paid for it.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
-Original Message-
From: john lawler [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 1:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] yet another question on DID trunks
Hey Steven,
Sorry to bother you yet again w/ a question on my seemingly endless
quest to
still will install/sell only T100P
cards and channelbanks to businesses, especially if
they will use VOIP at their location.
I believe the extra dollars you spend on
the equipment will more than make up for the time you will spend on
troubleshooting problems related to echos.
Tim Thompson
since you'll be able to
plug it directly into your * box. 8-)
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, January 05, 2004 10:36 AM
To: [EMAIL PROTECTED]
Subject: Re
script with if/then statements that would check to see
if the call was coming internal channel or external channel and then
would apply the Callback/Transfer code when appropriate.
I'll let someone else comment.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
and reloading.
And # is the Transfer button. 8-)
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Darren Nickerson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 9:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
if they can make any
sense of
this.
-Darren
--
Darren Nickerson
Senior Sales Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax
- Original Message -
From: Tim Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday
Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Tim Thompson
Sent: Wednesday, December 31, 2003 1:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone?
Have you tried deleting those services
480, Adtran 750 Channelbank (updated firmware), T100P
card, and it worked fine on the first try with current CVS.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Darren Nickerson [mailto:[EMAIL PROTECTED]
Sent: Monday, December 29
It has to do w/ your /etc/asterisk/Zapata.conf file.
We need to know what your T1 configuration is.
24 FXS ports
12 FXO Ports/12 FXS
and also it matters in what order.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: [EMAIL
1-24) and then there are 2 X100P FXO
ports (channels 25 26)
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Monday, December 29, 2003 8:56 AM
To: [EMAIL PROTECTED]
Subject: Re
Just cross pairs 1-2 and 4-5
1 -- 4
2 -- 5
Never done an E1, but I think it's the same for a T1 4 wire cable.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: bam [mailto:[EMAIL PROTECTED]
Sent: Monday, December 29, 2003 9
I've searched through the lists, but can't seem to find a reference to
someone that said they had a time and temperature application for *.
Voip-info wiki would be a good place for it as a Festival example!!!
Can someone remind me where it's at?
TIA.
Tim Thompson
You need to make sure you have a register
command in the iax.conf file.
[iax.conf]
register = username:password@iaxtel.com/extension
to send incoming calls to
I belive
username has to be all lowercase too.
Tim Thompson
Commercial Sales Engineer
http
and they would come in on
the FXS channels and the POTS lines would come in on the FXO channels.
In our area, Trunk lines run about $29-$35 each and then you pay for the
DID's.
Hope it helps.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From
Just a note to Mark and others.
In queue.conf, there is a reference to announce-markq that I believe
comes default uncommented.
There is no sample file in /var/lib/asterisk/sounds/announce-markq
If there is no file there and/or you misspell the filename and the
system can't find the announce
take the T1 right into your * box and you'll have plenty of
channels to spare in the future.
Tim Thompson
CPE Manager
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: John Vozza [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 10, 2003 7:54 PM
To: [EMAIL PROTECTED
I would get one of the X100P TDM400P
combos from Digium.
http://www.digium.com/index.php?menu=developerskit_tdm
If youve already got the computer,
youve got yourself a nice phone switch.
Tim Thompson
CPE Manager
http://www.amatechtel.com
(806) 722-2227
-Original
.
-Original Message-
From: Raymond McKay [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 8:27 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Forwarding a call to another FXO port
-- Original Message --
From: Tim Thompson [EMAIL PROTECTED
I don't know if anyone had used these boxes yet.
I've installed some, but not connecting to an * system. The guys who
ordered them said they ran about $600.
They are pretty cool in that you designate what the port will be either
Data, FXS, or FXO(not fully implemented yet)
This is a NEW price
I would change the option number to something else because 9 is often
picked up in another context as 9NXXNX
You might have to make a sub menu in order to get there, but try using
2-8 for the menu options.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
phones or POTS phones
for fax machines, regular phones, etc.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Howard White [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 12:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
= 0,2,Goto(default,s,1)
Tim Thompson
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: Todd Lieberman [mailto:[EMAIL PROTECTED]
Sent: Monday, November 24, 2003 9:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup
Yes.
Tim Thompson
http://www.amatechtel.com
(806) 722-2227
-Original Message-
From: CW_ASN - Gus [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 1:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Meetme
Do you have ztdummy or zaptel device in your system
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
Would anyone have an idea on how I would be able to take the mic in on
the computer and put it as the talking party for a conference room.
I would then be able to set up a listen only profile for others to get
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