,
*From:* Lyle Giese l...@lcrcomputer.net
*To:* asterisk-users@lists.digium.com
*Sent:* Tue, June 19, 2012 9:29:12 PM
*Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone
An FXO port needs to be connected to dial tone or your PSTN line. And an
FXS port needs to be connected
to talk
to the physical FXO port.
Lyle Giese
LCR Computer Services, Inc.
On 06/18/12 15:08, Joseph Towery wrote:
Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24
asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12
and asterisk-gui 2.1.0.rc1 (not trying to use
.
And one shield around all the pairs is not the same as ABAM.
Lyle Giese
LCR Computer Services, Inc.
On 12/08/11 10:53, Carlos Alvarez wrote:
A T1 cable according to this spec:
http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
Crossing the 1/2 to 4/5
at other
sites but this is the only site it is not working at.
Any ideas would be great.
Thanks,
*Kevin *
--
/var/log/mail on any of the SuSE or RedHat boxes I have looked at.
Lyle Giese
LCR Computer Services, Inc
. But with your option
turned on, they will know if they have a valid user name or not.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
...@lists.digium.com] On Behalf Of Lyle Giese
Sent: Friday, July 22, 2011 8:07 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] use dahdi for local terminal modem
access?
On 07/22/11 18:13, William Stillwell wrote:
I have some terminals that have phone lines.
One of my tech had an idea
needed to do was telnet
into an APC masterswitch to toggle power on one outlet. It worked.
I was surprised at getting a 14,400bps connect. I was not expecting
that high and really did not need that high. 300 baud probably would
have been fast enough to telnet into an APC masterswitch.
Lyle
the way you want
it to.
Lyle Giese
LCR Computer Services, Inc.
On 06/20/11 10:05, Sagbo Romaric wrote:
Ok, thanks,
Can you help me to have this kind of rules ?
I try with iptables without success.
Best,
Romaric SAGBO
*De
need an extension 811212.
I would use:
[inbound]
exten = 811212,1,answer
exten = 8151212,2,Goto(mainmenu,s,1)
exten = 811212,3,hangup
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation
Jonathan Hunter wrote:
On 24 November 2010 01:20, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Post the revelent portions of your extension.conf. Maybe you have
a logic error somewhere.
Thanks Lyle.
My extensions.conf is fairly simple in this regard; I use
the Zhone that tries to
'trip' the ringing.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Bruce B wrote:
Hi Everyone,
I have pfSense running which supplies Asterisk with DHCP. I had some
testing ports opened for a web server which I have totally closed now
but when I chose option 10 (filter log) on pfSense I get all of this
type of traffic (note that it was only 1 single IP and
. It might cost
us bandwidth for no reason. In fact there is no open ports on our
network whatsoever.
Thanks
On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Bruce B wrote:
Hi Everyone,
I have pfSense running which supplies Asterisk
Gilles wrote:
Hello
I'm sure someone has already tried this: I use a couple of electric
heaters to heat my office.
I'd like to somehow connect them to Asterisk so that I could switch
them on remotely by either calling the IVR or sending an e-mail to the
Asterisk host, so that the room is
not sound right...
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
, the nat device will drop the connection in it's
nat table and thus disconnecting the softphone from Asterisk. (after the
router's timeout period of course)
2) The other issue is you are connected to a conference call and you
want to mute your transmitter while listening to the conference.
Lyle Giese
://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip-sipsak/details
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Barry Fawthrop wrote:
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
.
Depending on your machine, I am guessing that Asterisk locked up or
dropped out on the 23rd and the restart on the 26th brought it back to life.
Nagios is a good choice for monitoring servers and services. I use it
here to monitor all the servers and SIP on my Asterisk box.
Lyle Giese
LCR
Lyle Giese wrote:
bruce bruce wrote:
I am not sure why it would be sleeping. I have never dealt with
putting a linux server to sleep. It is connected to a UPS, but I
don't think it has been put to sleep by the UPS as the USB cable from
UPS is not connected to it.
Can you please elaborate
Maybe you need to read the man page for qpage. The qpage client can
send the page to an SNPP server over TCP/IP.
Lyle
AMARDEEP SINGH wrote:
Our SMS-gateway is not PSTN accessible.
On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote
qpage -s snppserver.example.com -p lyle -f lyle test page
AMARDEEP SINGH wrote:
Do you have working script?
On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Maybe you need to read the man page for qpage. The qpage client can
send
AMARDEEP SINGH wrote:
Hello All,
Scenario:
-We use asterisk as voicemail server for our cellular network.
Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
-Extensions in * are virtual, just for leaving and accessing voicemail.
Requirement:
Asterisk to send SMS to
), or is the mechanism based on: I talk first and the sever
gets back to me based on that.
Should not need any forwards. However the router could be firewalling
some ports, like the rtp ports. You need to ask what ports are needed
for rtp.
Lyle Giese
LCR Computer Services, Inc
Tim Nelson wrote:
Greetings all-
I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks
rather interesting. Has anyone used one? Where did you purchase it? Pricing?
Operational issues?
http://spidermux.com/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
, check syslog-ng.conf and the summary option.
Setting summary to 0 turns off that behavior.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Patience is a virtue.
Demanding answers or responses is a sure fire way to get ignored, esp
since you waited only a few hours for a response. Here's it's Sunday.
Traffic levels are down over the weekend as most list users here are
doing family things instead of their jobs.
Besides, this list
Warren Selby wrote:
On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Here's a start for you, just run from cron once a day:
Lyle
So basically, nothing built into asterisk that already provides
security logging mechanisms? Maybe
Warren Selby wrote:
Hello list,
I've got a client who's weak sip passwords are being guessed by remote
entities who then connect to their server and use it to wardial large
swaths of numbers. When they start receiving complaints, they call me
and I add the ip address of the remote
that will include answer supervision.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
to connect
to the PSTN.
For instance if you are using POTS(plain old telephone service - analog
copper fed lines), you do not get answer supervision back from the telco.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth
Jeff LaCoursiere wrote:
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes
there support booting from a USB
drive, so why bother? Get one good DVD drive and put it in a case with a
USB adapter in it and just plug it in when you need it.
Lyle Giese
LCR Computer Services, Inc.
--
_
-- Bandwidth
, and typing dahdi show cadences in the CLI after
the restart showed my custom cadence, but the phones were still
ringing long ring-pause. Can someone point me in the direction of
what I'm doing wrong?
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
Lyle Giese
LCR Computer Services, Inc
on it, but whose TDM card are
you using now.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
have the following cron job:
/usr/sbin/asterisk -r -x 'restart when convenient'
Doug
You probably don't need the single or double quotes at all. I have never
used any quoting in crontab.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth
/[r...@dhcppc0 asterisk]# vi extensions.conf
[tutorial]
exten = 1234,1,Dial(SIP,gianca)/
/exten = 12345,1,Dial(SIP,giusy)
/
Here the XLITE user data:
/Display Name: gianca/
/Username: 1234/
/Password: pwd_gianca/
/Authorization User Name: 1234/
/Domain: 192.168.1.100/
Contexts.
Put the 'Source channels' in different contexts.
Lyle
B.Masoud @ SH wrote:
Can you tell me how on the first question?
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
to Asterisk and each can have their own
extension instead. It just requires cat 5 cable back to a switch for
each phone.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
B.Masoud @ SH wrote:
Hello,
I have :
answeronpolarityswitch=yes
on chan_dahdi.conf
but it's making all my lines answer on polarity reversal, this causes
a problem for PSTN lines, so how can I set these lines to answer
immediately (when it rings)?
thanks
Ishfaq Malik wrote:
Bumping this in the hope that it is seen by people who missed it before.
Ishfaq Malik wrote:
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk
server (1.4.17) as a SIP extension. This customer needs the dialled
number sent to the PBX as well as
Vincent wrote:
Hello
Out of curiosity, has someone managed to run Asterisk on a Beagleboard
for home-use?
www.beagleboard.org
As an alternative to a PC, it can be powered from a USB hub, so that
would make for a compact, fanless Asterisk server.
Thank you.
And now that the whole world of Asterisk has your sip user ids and
passwords, you should change all of the passwords that are in that file
and yes, change the passwords in all your phones.
Lyle Giese
LCR Computer Services, Inc.
hadi motamedi wrote:
Thank you for your reply . Please find
The receiving server does not ask for any user id or password. The
protocal says, the sender has to just send the user or pass command with
the data required.
Try reading /var/log/mail(if you have access), at least that's where the
outgoing mail logs on my servers are.
Lyle
Joan Antoni Terre
the proper account at Teliax
and you get the proper caller id set.
My inbound is still pots lines from the telco, btw. There is no
significant cost savings on inbound for telco vs VoIP here.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth
and I don't see anyone other than the phone company willing to spend the
money to make it happen.
To keep this on topic for Philipp's remark, the only bonus points we
assigned was to correctly guess how many phones were attached to the
phone lineGRIN!
Lyle Giese
LCR Computer Services, Inc
resolution very well.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
of the cable which is dependent on length and gauge.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Even with 'conventional' PBXs, there is such a thing as power fail
devices where the extension is cut to a telco pots line for dial tone if
the PBX goes down.
Jon Pounder wrote:
John Novack wrote:
If this is an emergency phone situation then I would question the wisdom
of even considering
Brent Vrieze wrote:
Lyle Giese wrote:
Manoj Panicker - FOES wrote:
Hi
Which is the best interface card to connect* PSTN* line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa
. Not the
cheapest way, but it has served me very well.
You are not going to get much help unless you define the problem better.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
Brent Vrieze wrote:
openSuse 11
Asterisk 1.4.23.1
Asterisk GUI 2.0
When parking a call it does not tell me what extension it parked the
call on.
I think I read something in the mail list that mentioned a problem with
call parking and one of the Asterisk 1.4s.
Is 1.4.23.1 one of those
A channel bank != PRI. A PRI is ISDN. A channel bank is not the same as
a Primary rate ISDN line.
With a channel bank, each channel's signaling is done in the channel.
Primary rate ISDN has a D channel to contain all signalling for the 23
voice channels, taking over the 24th voice channel.
Lyle
if nothing else?
Cary
this works here in my extensions.conf(with my fax line in this context):
[outonly]
exten = s,1,Wait,20 ; setup for fax line to stop ringing
exten = s,2,Hangup
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth
David @ULC wrote:
Lets presume that my both software are open. Xlute and Eyebeam
But I want my calls from Asterisk to land only on Eyebeam and Not on
xlite. How to set it ?
Give each their own SIP credentials. Then in Extensions.conf, when
dialing into your extension, send the call to both
David @ULC wrote:
In windows, we use BAT file to execute few series of command , which
help us in not writing each command manually everytime we want to
execute those commands.
In CentOS, I want to do the same thing.
Any Advice ?
Why? This is not an Asterisk problem...
You need to find a forum specific to your linux distro...
Lyle
David @ULC wrote:
Sorry to bump it , but any help ?
Like un-installing the driver and reinstalling it will solve the issue ?
Or shld I reinstall the OS again ?
On Sun, Jan 25, 2009 at
How many incoming calls will they support per line? You may find that
they support more than one incoming call per number.
Otherwise, get another provider.
Lyle
Alfred Monticello wrote:
I'm still stuck with this problem..Would appreciate any ideas anyone
might have on this one.
Thank you
,
could be CRC errors, could be no signal, could be ?
6 Blue alarm means I am receiving AIS or all ones signal, can be framed
or unframed.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Joseph wrote:
We have a caller ID from our phone provider Shaw Cable (digital phone) and
it was working OK until recently.
I get an error:
WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4,
digest has pstn-
NOTICE[6769]: chan_sip.c:14316 handle_request_invite:
If you are running the script within Asterisk as root, then it's a path
environment issue. My guess(and I run into this with cron jobs all the
time) is that the path is different from the command line than the
environment that the script runs under.
There are times where the fix is to use the
Gordon Henderson wrote:
On Thu, 8 Jan 2009, Thczv F. Thczv wrote:
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that
bala krishnan wrote:
Hi Friends,
Currently i am using the asterisk 1.4.x version. In that i want to
enable to silence suppression in the SIP calls. Please tell me the
configuration changes to be done.
Thanks in advance,
balasam.
Enabling silence suppression is a bad thing.
sasikala kala wrote:
Hi,
I have a requirement, whenever a user comes into the conference, it
has to announce the user name to all the person who are all available
in the conference.
I have used Meetme(,di)
where i is to announce the user leave/join with review.
I user used I also, which is
You need to implement SMTP-AUTH and log in when sending mail to your
smart host. I have a template for Postfix to do that. Many *nix distros
have Postfix with a sendmail compatible binary in front of it.
Lyle Giese
LCR Computer Services, Inc.
[EMAIL PROTECTED] wrote:
When I send email from my
T1 is NOT DSL. Most T1 links you purchase now are brought into your
building with a type of DSL conversion to extend the distance between
repeaters/amplifiers. T1 is purely a digital signal. DSL converts the
ones and zeros to audio(multiple tones to provide multi channels of
data). A
Brian J. Murrell wrote:
I'm looking into getting a new phone and wondering what the difference
in functionality is between a single line phone with call waiting and a
real 2 line phone (either a real SIP phone or an analog 2 line phone and
a 2 port ATA) is. Why would I want the real 2 lines
Brian J. Murrell wrote:
On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote:
1) a two line phone can register with two different * servers or sip
carriers.
Indeed. But if I only had the one * server which itself registered to
my carriers...
2) It's easy for both incoming
Dean Collins wrote:
Has anyone ever 'released' an Asterisk module that is easily
shared/downloadable?
Or doesn't the nagios open source code work like that?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent:
RoLaNd RoLaNd wrote:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to
be able to talk for more than 5 min at a time!
or like 20 min per week! or whtever
telephony
experience with Legacy systems.
Any help is appreciated.
Most likely the box is using sendmail or postfix to send those emails
out. You need to setup sendmail/postfix to use a smarthost using smtp
auth to allow relaying from this box.
Lyle Giese
LCR Computer Services, Inc
I bet the reason is that when his gf calls, he can erase the records so
his wife's divorce attorney can not get his hands on them to play in court.
Lyle
Eugen Soare wrote:
So basically,
He wants all calls recorded, but he wants a sequence that he can
push, so that when he rants and raves
Your E1 links are down. (red alarm) Your card does not like or see your
providers E1.
Lyle
Bikrish Amatya wrote:
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
will
tell you the response to it and you are in.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
to the
encrypted data.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Ronald Wiplinger wrote:
I have a local asterisk 1.2 and a remote asterisk 1.4.
Snom 190 can be used with the local asterisk but not with the remote one.
I need some hints where to track down this issue.
Some information:
Snom 190:
Line 1:
Account: 615
Password: OnlyIknowit
Jay R. Ashworth wrote:
On Tue, May 20, 2008 at 07:03:06PM -0500, Lyle Giese wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1
. If you buy new, I think they are around $1200 in the US.
And if you are not sure how to configure them, ADC is quite
accommodating and I have configured more than one and can assist.
Lyle Giese
LCR Computer Services, Inc.
___
-- Bandwidth and Colocation
Doug Lytle wrote:
Don Pobanz wrote:
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
This does not sound right. If it is 2 PRIs then it should be 46 channels
I may have the terminology incorrect. I don't have a D channel, so I
guess this would be
James Finstrom wrote:
Anyone have the telemarketer torture prompts? I would seriously like
to revive this.
Weasels and Monkeys work well for this.
I put up one extension that uses Monkeysintro then Monkeys and loops.
The other extension uses somethingwrong then weasels and again loops
back
on how to set Postfix
to use SMTP AUTH when sending email.
Lyle Giese
LCR Computer Services, Inc.
Mike Hammett wrote:
I am the ISP. ;-)
I'll have to look into that smarthost deal as there is no reverse DNS at
this time (my upstream's server times out).
--
Mike Hammett
Intelligent
If you take Asterisk down, the PRI should go down as the D channel is
down. Then the telco should KNOW that there is trouble with the PRI and
those channels are in trouble busy and not availible. If the telco
still tries to push a call to a channel on a PRI that is down, then the
telco is at
Why not give the receptionist a two line phone? Register one line on
server 1 and the other on server 2. Then the bounce back and forth goes
away saving bandwidth.
Lyle
Daniel Cole wrote:
Hello List,
I am currently having a bit of a strange issue with a pair of asterisk
servers that we
You need to do a 'make' before the 'make install'.
Lyle
[EMAIL PROTECTED] wrote:
Hi all,
Please help me in installing Asterisk.
I am getting the following error when trying to install Libpri
[EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2
[EMAIL PROTECTED] libpri-1.4.2]$ make clean
rm -f
daniele visaggio wrote:
2008/1/7, map [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put
a number greater than 200 :-). I suggest 10.
Sorry, i'm
Olle E Johansson wrote:
All I can say is with 1.6, if a change is made that causes something
that worked in 1.4 not to work in 1.6, please think twice, three
times or four times before making the change, or making the change
in such a way that it won't break dialplan stuff from 1.4.
Brian J. Murrell wrote:
On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote:
bump...
What's with all this bump I see here? Is this a web forum?
b.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Yep, that fixed it. Just shaking my head as to why the behavior changed...
Lyle
CunningPike wrote:
Try dtmfmode=inband
CP
Lyle Giese wrote:
I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13
with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix
config files
served up via tftp and only made the minimum required changes to config
files in Asterisk. I am running firmware 4.77(also tried downgrading
firmware on phones to 4.63).
Any suggestions?
Thanks,
Lyle Giese
___
--Bandwidth and Colocation
And why are you asking in the Asterisk list?
The absence of that file means you don't have any scsi adapters in your
system.
Lyle
Jarga Jallow wrote:
I am getting
this error under system info:
File
Line
The orginal did not make it to the list... Spam filter issue???
No repeat of the lockup yet.
Lyle
Original Message
Subject:voicemail locked up Asterisk 1.4.13
Date: Thu, 01 Nov 2007 20:57:27 -0500
From: Lyle Giese [EMAIL PROTECTED]
To: Asterisk Users Mailing
Mojo with Horan Company, LLC wrote:
Lyle Giese wrote:
Philipp Kempgen wrote:
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several
Philipp Kempgen wrote:
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things, but I think the correct
syntax is:
Set(_ALERT_INFO=http://127.0.0.1
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Lyle Giese
*Sent:* Friday, October 26, 2007 5:54 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Treating T1 as trunk in/out, not
individual
I am trying to get distinctive ringing going again with these phones,
depending on the outside line the call comes in on.
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things,
Your signalling is wrong.
The channels as programming in * should fxsks (use ks instead of ls) and
not fxols.
At Verizon's end, they use fxo and you grab it via fxs emulation in *.
Lyle
John Millican wrote:
Hello All,
I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not
Michelle Dupuis wrote:
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Thanks
Michelle Dupuis wrote:
I'm tying a Nortel option 61 to asterisk via T1. I don't want to
split each of the t1 channels out into individual lines (tied to a
specific extension) - so a trunk in and out.
Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER
info across the
Zaptel creates a startup script. You just need to make sure it run/loads
fully before Asterisk starts in your bootup scripts.
This gets into tweeking your system and that varies based on the exact
OS/distro you are running.
Lyle
Mojo with Horan Company, LLC wrote:
I don't have T1 but it seems
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe EM.
I have done EM wink but have no idea about E4 Superframe EM and Google
is not
1 - 100 of 263 matches
Mail list logo
