Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail password so
I am using Background(pls-enter-vm-password).
lst.ara [EMAIL PROTECTED] writes:
For our helpdesk application, we need record full conversation between
any caller and one or two helpdek numbers (while the conversation is
running). After conversation is ended (hangup ..), the recorded file
(WAW) is putted into database. Using AGI, record
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Maxim Litnitsky
Gesendet: Sonntag, 29. August 2004 23:59
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] SMS Asterisk
Hi all! I am intrested in the following scheme
My mobile phone - SMS to SOMETHING -
Hi,
I just got my zaptel fxo cards working, and I want to be able to
have someone call in on one line and access the other - I guess what I
want to do is transfer(exten), but that is only for extensions - not
channels which is what I want i guess. I tried the Dial(Zap/2) but I
think that's
I too have been playing with the cisco
7940's. I am guessing, as they only have two line appearence keys they
are only able to handle two lines. However they appear to be able
to handle 4 simultaneous calls (two per line key). If you assign two different
numbers, one to each key then you can only
Martin Holler napsal(a):
lst.ara [EMAIL PROTECTED] writes:
For our helpdesk application, we need record full conversation between
any caller and one or two helpdek numbers (while the conversation is
running). After conversation is ended (hangup ..), the recorded file
(WAW) is putted into
I do not get calldeflection (capiCD) to work.
The Mobile do not ring, it seems the CD do not work.
I use the chan_capi 0.3.5 and have no idea.
please help me.
nico
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
[...]
Why not send the sms to * directly?
It works in .de and .uk for sure.
[...]
Can you enlighten us as to how exactly?
Axel
--
Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany
VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265
Hello,
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
call my friend over the satellite - I call perfect but they cannot make out
a word I say. However if I leave him voicemail on his asterisk box, it
records my voice perfect. I have this problem when calling other
Hi,
Using a cisco 7960, if I try and transfer someone using
the transfer button, when I transfer them to a queue, it seems to disconnect
them. Does anyone know why?
I simply have an extension that points to a queue
(e.i. exten = 281,1,Queue(Sales) ).
Cheers,
Ben Merrills
Thanks, I was afraid to try and change gains that I didn't know what
they did simply because I don't want to blow a speaker or something.. :)
I'll try it today.
The only thing I haven't figured out is how to set a default ringer in
the configuration file, set my time to EST w/ Daylight Savings
Astricon is the first Asterisk user's and developer's conference to be
held in Atlanda, GA, USA Sept 22-24 this fall. Astricon is organized
by Edvina.net and Sokol Associates in partnership with Digium, inc.
** CURRENT ESTIMATE: OVER 250 ATTENDEES
Steven and I started this project at the end of
Opps, at 3am I make stupid editing mistakes. Should read:
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
call my friend over the satellite - **I can hear him perfect**, but he
cannot make out
a word I say. However if I leave him voicemail on his asterisk box, it
Hi,
If I try to transfer a user (user listens to MOH while I transfer) to eg. a
queue, and the transfer occour while the MOH in the queue is playing,
the MOH will stop, and the user hears nothing but scilence, but is in
the queue.
If I make the transfer to the queue, while still listening to
Hi there,
this is just a me too... well, not exactly. I get jitter when trying
to make SIP calls through Asterisk using a GPRS connection... can this
be done actually?
TIA,
Martin
Storm D. J. Petersen wrote:
Hello,
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
Enrico Stahn wrote:
Hi!
Have a look at the following entry. I solved this problem:
http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error
That's the wrong way to do it. You use incorrect versions of
the libraries.
Michael.
___
Asterisk-Users
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as it should be matched by the first pattern. Any
Hi,
-Original Message-
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers,
where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth
pattern, where
as it should
On Tue, 2004-08-31 at 21:42, Atif Rasheed wrote:
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as
You did not replace the existing prompt, but added a second prompt. The
proper place to make this adjustment is in VoicemailMain, not in extensions.
Or find the password prompt sound file and just replace it with yours.
Lyle
- Original Message -
From: Java Rockx [EMAIL PROTECTED]
To:
On Tuesday 31 August 2004 11:36, Martin Mielke wrote:
Hi there,
this is just a me too... well, not exactly. I get jitter when trying
to make SIP calls through Asterisk using a GPRS connection... can this
be done actually?
[...]
Yes, we've done it over Vodaphone (I think). The lag,
about 1.5s
Hi,
I'm trying to answer a call on one line and dial out a number on
a zaptel x100p fxo, but all I get from the phone I'm dialing is silence
after it is picked up, and on the line that's supposed to be dialed out
itself, noise.
Thanks,
Imran
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your macro remove the line that plays the
I don't mind latency ... it's the garbage jitter where no one can understand
a word.
Interestingly enough if I do this it works fine:
[grandstream 1]- [sat]- [pbx in mothers house]
[grandstream 2]- [sat] -/
where the grandstream phones are side by side.
S.
-Original Message-
From:
Chris Shaw wrote:
- Channel Support:
IAX2 in asterisk
IAX2 in libiax2
Other IP channels in asterisk (RTP-based ones, I guess are all that is
left).
CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete
solution... As much as we all hate it's complexity and wish
thank you people for your help, I have done it, and in a different way,
like
exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011X.,1,AGI(iax.agi)
exten =
Good day all
I'm new to the whole pbx thing.I've setup 2 servers with voicetronix card!
Each card's got 4 ports.Ive configured it so each port is for a different
company,so in other words if a call comes in on port 1 it plays company 1's
welcome message ens..I did this with context in vpb.conf
Pick up mobile phone.. enter sms .. send it to the * phone number.
Done
On the * side.. follow the sms howto (voip-info.org might have some infos)
Done
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Axel Eble
Gesendet:
On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke
[EMAIL PROTECTED] wrote:
Pick up mobile phone.. enter sms .. send it to the * phone number.
Done
On the * side.. follow the sms howto (voip-info.org might have some infos)
Done
Ah. That requires SMS to be available on land lines.
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Axel Eble
Gesendet: Dienstag, 31. August 2004 15:35
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: AW: AW: [Asterisk-Users] SMS Asterisk
On Tue, 31 Aug 2004 15:22:26 +0200, Michael
Hi,
I'm trying to configure a natted budgetone
phone to a asterisk server as described in wiki using port forwarding.
I successfully make call from the client
but it seems it does not register the client ip address and when I try
to recall it is not reacheable.
Asterisk can manage natted sip
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine,
Hi
I want to play a bit with Asterisk. I currentlly install a new system
for that and I would like to get your recommendations regarding the
linux distro to use there.
This is NOT intended to become a general distro flame war. My favorite
distro is and no argument that you flame will
Here's my iax.conf and extensions.conf (I have not yet made the recent
changes they just emailed about a day ago, this is twice in a two month
period, jeesh) I have tested inbound and outbound dtmf. I use the g.711
codec and use inband.
iax.conf
Hi,
I have this strange problem I need some help with.. It appears that I have
harddisk noise captured by a Digium TE410P card (Same problem on 2 identical
machines..) The machines are two Compaq Proliant DL320 G3's...
Does anyone else have this problem..
Kind Regards
Claus Futtrup
---
We discussed this earlier and I believe the general consensus was that
it's personal choice. I've personally used Asterisk on Redhat 9.0,
Fedora Core 1 and Gentoo 2004.2
Each has required some minor securing and cleaning up, but Redhat/Fedora
tended to need more babying as far as securing
You can actually hear the hard drive noise when calling out or receiving
a call? A clicking sound, or like an electrical noise?
I doubt this is being done through the motherboard, how close is the
card to the power supply and/or the power wires going into the hard
drives? Are they less (or
What does your host= line show in the iax.conf for fwd? I found that
iax.conf hates it when you use host=x.x.x.x so instead I had to use
host=dynamic and defaultip=x.x.x.x or something like that. It's very
finicky.
Storm D. J. Petersen wrote:
Hi,
I cannot seem to accept incoming calls from
Hi,
sorry for annoying question;
i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup
without understanding:
1. how to add an ext. to a pickup group (ie:. how to populate pickup group)
2. how 'Directed pickup' does work?
You dial the pickup number and your extension, and the
Asalamualaikum Atif,
i saw your guy's ad in spider magazine. sounds cool... yeah, i got
asterisk to work, i had to build zaptel before asterisk. just trying
to transfer from one line to another now...
thanks
Imran
Atif Rasheed wrote:
well Imran, I am not a guru of Asterisk, but I think
I'm using it on Debian Stable (Woody), works great, using it with the
backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with
their headers. I think it's all a matter of personal preference. I
prefer Debian, so I use it, use whatever you like best :)
-Tim
-Original
I'm using RC2 and last weekend's changes from VoicePulse. Outbound
calling and dtmf works fine. However, an inbound call to my DID cannot
send dtmf digits to the IVR.
Thoughts?
Deon Rodden wrote:
Here's my iax.conf and extensions.conf (I have not yet made the recent
changes they just
I've been having troubles compiling in the openh323 on both redhat and
debian... one of the biggest problems I had w/ Debian is it couldn't find
alot of libraries like termcap etc...
Has anyone else ran into these problems?
-Original Message-
From: Tim Jackson [mailto:[EMAIL PROTECTED]
Hello fellow * users,
I've been experimenting like a madman lately with asterisk, and I just
love it. Just reading this list
and asking a few questions here and there has helped me out a great
amount. Not to mention the
excellent resource we call the Wiki.
I have searched for answers to the two
I had that problem, but apt-get install did the trick.
On Tuesday 31 August 2004 02:53 pm, Huddleston, Robert wrote:
I've been having troubles compiling in the openh323 on both redhat and
debian... one of the biggest problems I had w/ Debian is it couldn't find
alot of libraries like termcap
- Original Message -
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail
password so
I am
On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote:
Hmmm...
Hands Free might be:
voice.gain.rx.digital.chassis=15 (15 is my setting)
Call waiting? You can turn it off in sip.cfg - do not disturb settings
I think. Don't know about gain for call waiting. You might try playing
with some of
On Tue, Aug 31, 2004 at 11:02:30AM +, Brian Wilkins wrote:
I had that problem, but apt-get install did the trick.
Not to mention apt-get source and apt-get build-dep if you need to patch
existing packages
--
Tzafrir Cohen +---+
1) The samples are empty? No, they have variables with settings. Maybe
I'm not understanding you.
2) I don't know how to dump the current settings to an xml file. You
might try increasing the log level, but I doubt you're going to get a
pretty looking xml file written to the log files.
I got most of the features of my phone working. Polycom TEch support
refuses to help or even talk to me. So I'll have to ask here again.
On incoming calls, only the NAME is displayed. I am trying to figure
out how to get the NAME NUMBER displayed.
If anyone can help me do this it would be
If that was possible, that would make my life easier as well :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Tuesday, August 31, 2004 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
John,
By chance do you know how to set a default ringer?
What I have done is the following:
DEFAULT se.rt.1.name=Default se.rt.1.type=ring
se.rt.1.ringer=7 se.rt.1.callWait=6 se.rt.1.mod=1/
As you can see, I want 7 to be the default ringer for line 1... For some
reason, it doesn't
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue,
Is there a solution for asterisk to send the calling costs
to a display of a grandstream Bt101 phone.
Does anybody know if there is a solution for this?
Greetings Han
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi everyone,
I'm having a little problem and was wondering whether anyone would have
any ideas or pointers for me.
I've been working on load-balancing asterisk and have had a pretty
successful setup using LVS and IP tunneling (plus a bit of iptables
nating).
I am only load balancing the SIP
I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
fine on my 7960... W/ the name on top and the number below that.
-- Executing NoOp(SIP/614-3ede, Caller*ID is Matthew Marlowe
6092521155) in new stack
When the phone rings, only 'Matthew Marlowe' would display. When I
On Tue, 2004-08-31 at 10:37, Matthew Marlowe wrote:
I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
fine on my 7960... W/ the name on top and the number below that.
-- Executing NoOp(SIP/614-3ede, Caller*ID is Matthew Marlowe
6092521155) in new stack
When the
Claus-
This is a problem that interests me, as I'm about to deploy TEN of these at
a customer site, all with TE410P's.
I'm currently load testing one Proliant box (3GHz P4 processor) looping 59
calls out to 59 calls in (leaving one channel open) - ie: lots of load.
While I'm doing this, I call
This is nothing to do with SIP. It is an RTP issue, common to everything
which uses RTP - SIP and H.323 included.
I have been reading the RFCs and I'm a bit more familiar with how it works
now although the algorithms are a bit over my head. I am somewhat new to
RTP/VoIP, but I have a strong
I'm using RC2 and last weekend's changes from VoicePulse. Outbound
calling and dtmf works fine. However, an inbound call to my DID
cannot
send dtmf digits to the IVR.
Thoughts?
I have the same problem...my iax.conf is set up exactly as recommended
per the recent Voicepulse changes and
I have been reading the RFCs and I'm a bit more familiar with how it works
now although the algorithms are a bit over my head. I am somewhat new to
RTP/VoIP, but I have a strong telecom/networking background so it makes
things a bit easier to understand since they share a lot of common
On Tue, Aug 31, 2004 at 10:15:02AM -0400, Deon Rodden wrote:
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,3,Congestion
This would dial the number twice..?
My config is
exten =
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of
After reading a retarded amount of docs I'm still unable to figure out how to
get Asterisk to monitor my phone line and pick it up when the phone
rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to
another stack of docs? Is this even doable without special hardware?
TIA,
My DID is 303 as well.
Marty Mastera wrote:
I'm using RC2 and last weekend's changes from VoicePulse. Outbound
calling and dtmf works fine. However, an inbound call to my DID
cannot
send dtmf digits to the IVR.
Thoughts?
I have the same problem...my iax.conf is set up exactly as recommended
If I put my outbound rules in a different context, and then include
them in my main context, callers who call in will be able to access the
extensions in the main context, but not the included (ie the outbound
extensions) extensions called from the outbound context?
Lyle Giese wrote:
You limit
Hi there,
The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of
ram., and the server is running Red Hat 9.0.
The sound is just like hearing a disk just muffled (sounds like strange
static)..
If you have a number I can call you at then you can hear it yourself.
Kind Regards
After reading a retarded amount of docs I'm still unable to figure out how to
get Asterisk to monitor my phone line and pick it up when the phone
rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to
another stack of docs? Is this even doable without special hardware?
No,
Claus:
One difference is that I'm using the slower ATA disk, not the SCSI.
Is the noise rhythmic (periodic) or constant? If periodic, what is the time
between noise bursts?
Do you hear the noise throughout a call, or just occasionally?
Regards
Scott Stingel
Scott M. Stingel
President,
Thank you!
I took your advise and replaced the original vm-password.gsm file. Worked like
a charm.
Thanks again,
Paul
--- Jason Kawakami [EMAIL PROTECTED] wrote:
- Original Message -
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
You should be able to do that, but of course always test, test, test to make
sure.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 11:24 AM
Subject: Re:
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten = you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to
We have a Dlink DVG-1120M and were surprised that it was able to handle 2
simultaneous conversations to 2 seperate phones using only 1 MAC address and
1 IP address.
So we asked ourselves..why can't other 1 MAC/1IP devices do this as well?
I have a Grandstream 486 that has 1IP and 1MAC. But I
It's very possible that the Polycom IP600 will work with this. As it is
just an implementation of a SIP standard for subscribing to the state of
other extensions.
As for the feature improvements you might see them from me, but not very
likely. It is easier for me to train my customers to hit *8
The HT486 is a single-line device with a PSTN pass-thru. The only multiline
IADs I know of are the SIPURAs and the Cisco ATA-186...
What you do is you create 2 contexts, 1 for each line of the device and you
set the host name to the IP address (or host name if applicable) of the IAD.
Set the
Hi,
sorry to bother you, but i need to connect 8 standard analog lines to 2
asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after
let this 2 systems to interact between them.
I was thinking to use the TDM400 card equipped with 4 FXO modules on both
sides.
Is it correct to
Hi,
I installed asterisk-addons and configured it so that the cdr is done on a mysql
database. Everything was fine, until I originated outgoing calls using the
manager API. The call itself is performed perfectly, but when I hangup, I get
the following warning on asterisk CLI:
Aug 31 14:29:23
We have a Dlink DVG-1120M and were surprised that it was able to handle 2
simultaneous conversations to 2 seperate phones using only 1 MAC address and
1 IP address.
So we asked ourselves..why can't other 1 MAC/1IP devices do this as well?
I have a Grandstream 486 that has 1IP and 1MAC.
Hi,
suppose I have agents waiting on a queue and I configure asterisk to dial out
and to forward the call to the first agent enqueued. Asterisk will do it even if
the answer to the call is busy.
Is it possible to configure asterisk to detect the busy signal and, in that
case, dial another
Hello all,
I'm working an a switchboard console for Asterisk
and would like to investigate using IAX Client library to Asterisk. I
don't seem to be able to find the source. I'm planning on a Win32
app. Guidance on where the source isor who to "take" to is
requested.
Jon
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call
AFAIK, this is not possible - but I'll throw it out there anyhow...
I subscribe to telco voicemail, for the event that all my pstn lines are
in use.
Telco gives me a stutter-tone dialtone when I have a message waiting.
Can a Zap card detect this stutter-tone and perform some action?
I'm using
[EMAIL PROTECTED] wrote:
Hi,
suppose I have agents waiting on a queue and I configure asterisk to
dial out
and to forward the call to the first agent enqueued. Asterisk will do
it even if
the answer to the call is busy.
Is it possible to configure asterisk to detect the busy signal and,
Nope, Asterisk will not do this, at least not without some serious
busy-detect action going on and some tinkering with the dial and agents
code, in which case any call that is not busy will have to wait a second or
two for Asterisk to say that it isn't busy.
Another way to go is to look into what
Hi,
I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall
(all ports we're set according to instructions) on DSL line.
When pbx connects to Digium demo server( I'm located in Slovenia, Europe),
it gets first few words, then silence and then comes back when enumerating
dial
On Wed, 2004-09-01 at 04:39, Deon Rodden wrote:
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I
Hi,
I'd like to implement scenario to send user to operator's queue by default
(if doesn't dial any extension) but only if there is operator agent logged,
so user could get response. If not I'd like to send it to voicemail...
Any quick advice ?
Thanks in advance,
Robert.
Sorry, I know it's OT, but does anyone know of a relatively inexpensive
headset that is compatible with the Cisco 7960?
I've tried the headset off Norstar phones, doesn't seem to work with or
without the amp.
| nate carlson
I don't know that Plantronics stuff qualifies as inexpensive but I have
been using Plantronics H headsets with the adapter at this link.
http://store.yahoo.com/founderstelecom/dirconcabfor.html
I have two of these cables and they work very well.
Bryan
Nate Carlson wrote:
Sorry, I know it's OT,
Hi there
Background:
I want to add DDI and voicemail to users on an existing analogue pabx..
It does not support ISDN.
I have 10 DDI numbers via IAX which I am having sent to my Asterisk
box. I have 2 X100P cards connected to 2 analogue extension ports of my
main legacy analogue pabx. I have
I found the same with lots of headsets and my 7940, but I've just
plugged the headset from my Norstar system into the *handset* port on my
and it works perfectly. It's not ideal but it'll do for now!
Cheers,
Benjamin
Nate Carlson wrote:
I've tried the headset off Norstar phones, doesn't seem to
Cisco headset pinout is different from normal ones (grr)
If it's just for you, (ie nothing too professional ;) you can snip the lead
of an existing plantronics type headset and do some reordering - this will
give you the necessary info (sorry - can't remember exactly how I did it):
Hi
Last week I installed Asterisk (release1) with digium t100p single span T1
(wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is
currently the only user of the t1 line. All worked well for about a half a
day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I
GN-Netcom has a nice little headset for about US $120. As to the
pin-out,
I believe that the headset port uses pins 14 instead of 23.
Dan
-Original Message-
From: Edward Eastman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 1:29 PM
To: 'Asterisk Users Mailing List -
Started a Wiki page here:
http://www.voip-info.org/wiki-Cisco+Phone+Headsets
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Edward Eastman
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, August 31, 2004 10:28
AM
Zaptel.conf sets t100p to be the primary sync source for the only span, as
suggested by many Asterisk users.
I'm trying to understand so please bear with me... The T100P is connected
directly to the Mitel? Or to the Telco through a T1?
What I mean is are calls coming into the Mitel from the
I use a Plantronics Supra H51 plugged straight into the headset port, and it
works great.
B. J.
-Original Message-
From: Nate Carlson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 15:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] OT: Headset for Cisco 7960?
Sorry, I
Hi Chris,
thanks for taking time to look this over.
T100P/* is connected to the Mitel IP-PBX/CU and it to telco - so I think our
setting is correct.
BTW, I did try 0 (as well as 2) without success, just for fun, before I
came on a good explanation
of the sync source in this forum.
Hi,
Can anyone recommend a BRI card which works fine with asterisk and which
supports point-to-point mode? Software fax detection should also work.
Price does not matter. :)
Digium seems to sell only PRI cards, and the Beronet drivers for
the quad BRI cards seem to be in an early stage of
Inquiring (management) minds want to know. I'm assuming it's because 'it's
funny how simple it really is to write a really decent voicemail system'?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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