From the Wiki:
Presentation indicator (octet 3a)
Bits
7 6 Meaning
0 0 Presentation allowed
0 1 Presentation restricted
1 0 Number not available due to interworking
1 1 Reserved
Screening indicator (octet 3a)
Bits
2 1 Meaning
0 0 User-provided, not screened
0 1 User-provided, verified and passed
1
On Sat, 20 Nov 2004, Brian Roy wrote:
I would look at putting a dual monitor on her desk. You can pick up a
15 flat panel and a video card for about the same cost as the SNOM.
Not to mention, you get quite a bit more benifite from the FOP
controls than you do busy lamp fields. It's a a new
You should always design an interface around a human being. A hard
I could not agree more. Usability is my focus in any software
system...including open-source, where it is typically the last thing
considered.
Greg
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On 12:08 AM 11/21/2004, Chris A. Icide wrote:
Okay, ignore the previous question, I figured it out.
for anyone else who may also not completely grasp the wiki explanation:
number is a octet, and the only bits you need worry about are bits 1,2,6 and 7
bits 1 and 2 define the screening indicator,
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other options please?
Mike Dent wrote:
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other
Hello ,
When comedian mail prompts for login info , no matter what I dial on the phone , nothing is sent to * . I'm using a budgetone 102 , with the latest firmware (1.0.5.16). I have set dtmfmode=Info in sip.conf.
I'm not sure if its the phone or * that is the issue. Any assistance would be
Gregory Junker [EMAIL PROTECTED] wrote:
$400-500 device here. Not very price competitive. I would like to
see less than half that.
I agree that any touch screen ought to be able to do normal computer
graphics. At this point, you are into normal LCD displays with touch
capability, which I
On Sunday 21 November 2004 11:16, James H. Thompson wrote:
Gregory Junker [EMAIL PROTECTED] wrote:
$400-500 device here. Not very price competitive. I would like to
see less than half that.
I agree that any touch screen ought to be able to do normal computer
graphics. At this point, you
Mike
I use Asterisk at home and have bought a couple of HandyTones ATAs. The
DECT phones are plugged in and work really well. The ATAs are £56 from
Goods2World (though the additional one I've just bought didn't work and is
being returned) and about the same from VoIPTalk (who are out of stock
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other options please?
Hello ,
When comedian mail prompts for login info , no matter what I dial on
the phone , nothing is sent to * . I'm using a budgetone 102 , with the
latest firmware (1.0.5.16). I have set dtmfmode=Info in sip.conf.
I'm not sure if its the phone or * that is the issue. Any assistance
would be
I'm able to get to VoicemailMain , however when I am there , when I
dial any digits for username , password , they are not registering , so
its like I am not dialling any digits at all.
Regards ,
JK
On 21/11/2004, at 11:11 PM, Clive Carter wrote:
Hello ,
When comedian mail prompts for login
Hi,
in the Snom FAQ I found the following information:
After staring up, the phone tries the URL given in the Setting
URL of the phone. ... BTW this setting can also be set via DHCP.
option tftp-server-name http://192.168.0.9/snom200{mac}.htm;
The documents used:
FAQ-04-06-14-sf.pdf
Problem sorted , needed to have send dtmf via SIP info in the budgetone
setup as well as in sip.conf.
JK
On 21/11/2004, at 11:18 PM, John Khina wrote:
I'm able to get to VoicemailMain , however when I am there , when I
dial any digits for username , password , they are not registering ,
so its
Saturday, November 20, 2004, 7:03:53 PM, Steven wrote:
SC On Sat, 2004-11-20 at 18:48 +0100, Tamas J wrote:
Hello!
I would like to know wether it is possible to have end-to-end codec
negotiation in iax2?
What I mean is...
In case the user dials a number available through PSTN, let's force
I am using the same idea.
But, you don't want to put {mac} in the file name.
Just use snom200.htm.
What the phone does, it first reads snom200.htm
and then automatically proceeds to read a file of form
snom200-000413xx.htm
Put lines for all phones in snom200.htm and the rest in the file
My problem is that I'm trying to do a flash on an active ZAP channel
to transfer a call, but every time the flash is performed the caller
that im trying to transfer gets disconnected. Here is a longer
explanation of whats going on.
I have a situation where I am linking asterisk upto a PABX via
Hi all,
I don't know why but my * is not accepting Register Messages.
Have you seen this kind of problem before??
I need help!
Thank in advance.
*CLI Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869 register_verify: Peer
'111' is trying
I don't know why but my * is not accepting Register Messages.
Have you seen this kind of problem before??
I need help!
Thank in advance.
*CLI Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869
register_verify: Peer
'111' is trying
I'm trying to connect * server from diax 0.9.8c client and * outputs this
errors on CLI
Nov 21 18:59:59 NOTICE[7316]: chan_iax2.c:5742 socket_read: Rejected connect
attempt from 192.168.0.4, requested/capability 0x2/0x2 incompatible with our
capability 0x400.
Reid A. Forrest wrote:
I don't know why but my * is not accepting Register Messages.
Have you seen this kind of problem before??
I need help!
Thank in advance.
*CLI Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869
register_verify: Peer
Michael Welter wrote:
snip
Channel: Zap/g2/3036701917
MaxRetries: 1000
RetryTime: 60
WaitTime: 45
Application: TxFAX
Data: filename.tiff|caller
Note: All calls are going to the same fax machine, so some attempts on
the second line will get a busy signal (there are two POTS lines in
group 2).
Hi,
Me and another guy are working on LCD drivers etc for Linux. The thing
is, the display would be run from your Asterisk Server. I.E. It will
need to be run from either Parallel, Serial or USB port. We will open
source it once finished, and are not too far off, probably just a spare
day
Hello,
The problem with the call files is that the busy tone is not being
detected, and the reason the busy tone is not detected is because the
fax tone (CNG) is being injected onto the line by the TxFax application.
When I remove |caller from the call files (no CNG tones), all fax
calls
First of all, I'm (and many are) sick to see blah blah
blah doesn't work with blah blah blah.
This is * ML, not nufone, not any other provider.
I was trying to be the more explanatory possible ...
Any way the issue was solved ...it was (of course)
a config conflict on extensions.conf
My
Nicolás Gudiño wrote:
Hello,
The problem with the call files is that the busy tone is not being
detected, and the reason the busy tone is not detected is because the
fax tone (CNG) is being injected onto the line by the TxFax application.
When I remove |caller from the call files (no CNG
I setup a basic extension with Playback(queue-youarenext) and it worked
perfectly, as planned.However, it does not get played during the queue
itself. See below: I did the Playback of queue-youarenext before dumping
the call into the queue, where queue-youarenext failed.
-- Executing
Time to reboot and re-start Asterisk, well, hrrm, monthly, news.
It's been a hectic fall with a lot to do, both before and
after Astricon.
At this time, we're preparing for two Astricon shows in 2005. And no,
we haven't made a decision on where to run the European Astricon,
not yet.
I am preparing
Hi list!
My skinny phone can make outgoing calls but incoming calls just keep
ringing for the calling end but the phone that actually should ring
doesn't ring at all.
I guess I have something messed up with the dial command.
I have this in skinny.conf:
[z4040]
device=SEP (actual
Hello all,
I didnt see this on the list so check it out if you
have a budgetone.
http://www.grandstream.com/Firmware/ringtone/music-ring-tone-generator.zip
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Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
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On Sunday 21 November 2004 11:42 am, Gregory Junker wrote:
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
Another strong possibility is that
Another strong possibility is that after a while, few operators would be
willing to continue holding their arms in the air to operate a touch screen.
Why would they be holding their arms in the air? You mount the touch
panel in the same place at the same angle as the current console...
Greg
Hi,
Whilst trying to get this Tecom phone working with Asterisk, it seems
to be unable to login. Using the 'sip debug' command from the CLI does
not produce any
output even though the debug of the phone shows it trying to login every
second or so?
The phone seems to be based on a Centrality
Andy Rosen wrote:
I setup a basic extension with Playback(queue-youarenext) and it worked
perfectly, as planned.However, it does not get played during the
queue itself. See below: I did the Playback of queue-youarenext before
dumping the call into the queue, where queue-youarenext failed.
If I want to use IAX instead of SIP, do I need to get gateway that
support IAX.
Are there such gateways?
I plan to connect 3 to 4 standard phones via gateway with *
In addition I don't want to use SIP to setup VoIP. IAX is more suitable
for communication over firewall.
--
#Joseph
On Sunday 21 November 2004 11:42 am, Gregory Junker wrote:
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
Another strong
On Sunday 21 November 2004 11:50 am, Gregory Junker wrote:
Another strong possibility is that after a while, few operators would be
willing to continue holding their arms in the air to operate a touch
screen.
Why would they be holding their arms in the air? You mount the touch
panel in
When starting asterisk -r, I see the following:
Asterisk CVS-v1-0-10/21/04-18:23:13, Copyright (C) 1999-2004 Digium.
Thanks for your help on this
Andy
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The
Andy Rosen wrote:
When starting asterisk -r, I see the following:
Asterisk CVS-v1-0-10/21/04-18:23:13, Copyright (C) 1999-2004 Digium.
Thanks for your help on this
The only thing I can suggest is to try an upgrade... I've looked through
the code, and app_queue uses the same calls to play
Hello,
I had the same problem with the SipTone - it's just a matter of
setting the dtmfmode in the sip.conf file.
I think I set it to inband - I remember setting it to either that
or rfc2833 or whatever that rfc number is - the correct number is
available in the sip.conf fdile itslf. Just fiddle
Hello Miguel
Thanks for this suggestion, but if the user has onlye a Grandstream
SIP phone on the other end, no PC, nothing, just the SIP phone. It can
be possible any encription in this case?
Fach
On Sat, 20 Nov 2004 11:51:46 -0800 (PST), Miguel Ruiz Velasco Sobrino
[EMAIL PROTECTED] wrote:
Hello list
We're trying to get a HFC card running in NT mode. Zaphfc loads fine as
module and ztcfg -v is run only once per reboot. Cabling is ok, the
ISDN-TE gets ist power fine. The HFC card does not share any interupts.
After the TE picks up the console (pri debug enabled) shows:
Nov 21
There has to be a router or switch to plug the phone into or the phone wont
be of much use.
You can pick up a cheap linksys IPSec VPN endpoint for about $80 last I
checked.
Hello Miguel
Thanks for this suggestion, but if the user has onlye a Grandstream
SIP phone on the other end, no PC,
Thanks Steve
Now with the help of all of you the picture is getting more clear to
present it, at least for me
Thanks for all your tips
On Sun, 21 Nov 2004 13:02:53 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
There has to be a router or switch to plug the phone into or the phone wont
be of
New to Asterisk so I
am sure this has been answered before. I can compile PWLIB and OpenH323 but when
it comes to compiling asterisk-oh323 then I get all kinds of errors even though
I have set the paths up in the source files. I can attach the errors if it is
useful. I though however that
I have a bit of a weird problem that I'm having great trouble debugging.
I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules
are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID,
which works fine with Asterisk. Both asterisk and zaptel are fresh from
Ok.Rebuilt using CVS, so I'm at:
Asterisk CVS-HEAD-11/21/04-12:45:30, Copyright (C) 1999-2004 Digium.
Still the same messages:
Nov 21 13:06:10 WARNING[13842]: file.c:475 ast_openstream: File
queue-youarenext does not exist in any format
Nov 21 13:06:10 WARNING[13842]: file.c:779
Andy Rosen wrote:
Ok.Rebuilt using CVS, so I'm at:
Asterisk CVS-HEAD-11/21/04-12:45:30, Copyright (C) 1999-2004 Digium.
Still the same messages:
Nov 21 13:06:10 WARNING[13842]: file.c:475 ast_openstream: File
queue-youarenext does not exist in any format
Nov 21 13:06:10 WARNING[13842]:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
Was wondering
- does anybody have experience with TDMoE over bonded interface - ie. does
it work ok?.
- does anybody have
hmm.. All the security stickers are still in place so I'm wondering
how that happened. The part number ( 2201-06622-001 A) is the number
for an Alcatel (the h323 version) phone, but it is booting up with the
3COM graphic and asking for the NBU server. From my research, the 3COM
is a
Great Suggestion!
In queues.conf, I had the following:
queue-youarenext = queue-youarenext ; (You are now first in line.)
queue-thereare = queue-thereare ; (There are)
queue-callswaiting = queue-callswaiting ; (calls waiting.)
queue-holdtime = queue-holdtime ; (The current est. holdtime is)
Andy Rosen wrote:
I found that Asterisk was looking for queue-youarenext- Ie: the
quotes shouldn't be there.
So, I took the quotes out of queues.conf and all is working!
I appreciate your suggestions. I should have dove straight into file.c
before initially posting, but...I'm glad you
hello,
i need some suggestion how to indicate caller that calling number is
unavailable if some iax user is not registered:
this is what I got in asterisk console:
app_dial.c:727 dial_exec: Unable to create channel of type 'IAX2'
== Everyone is busy/congested at this time
why is send to busy
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The kind of functionality that is being described here is one or
both of those 'other' beasts. Now I'm not saying that this wouldn't be
nice, or even a long term requirement if you really want to open the
entire SME
Andy Rosen wrote:
In queues.conf, I had the following:
queue-youarenext = queue-youarenext ; (You are now first in line.)
[SNIP]
Well, upon putting the following on 1144:file.c
ast_log(LOG_WARNING, %s\n,filename);
I found that Asterisk was looking for queue-youarenext- Ie: the
quotes
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Greg
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[EMAIL PROTECTED]
I wrote:
I've got some SIP clients, and an IAX2 long distance provider.
Ideally, when a the dialed number is busy I will hear a busy signal.
Instead, I get Congestion even though * knows it's busy. Is this a
bug or am I missing something?
Okay, well failing a response can anyone let me know
On Fri, 19 Nov 2004, Michael Van Donselaar wrote:
I think that iaxComm is currently the only other iax softphone for linux
http://iaxclient.sourceforge.net/iaxcomm/index.html
Thanks for the lead. I gave it a try on SuSE 8.1, and it failed with a
library incompatibility. I tried it on SuSE
Ok, so I realised I was running a CVS version of * which might have been giving
me the SIP problems. So I decided to get down 1.0.2. I followed the usual
instructions, compiled and installed it. (FC2)
Now when I try to start * I get:-
[chan_zap.so]Nov 21 20:37:05 WARNING[-150101888]:
Bill Seddon [EMAIL PROTECTED] wrote:
I use Asterisk at home and have bought a couple of HandyTones ATAs. The
DECT phones are plugged in and work really well. The ATAs are £56 from
Goods2World (though the additional one I've just bought didn't work and is
being returned) and about the same
Peter-
I haven't retried this lately, but it worked fine in the past when I did.
Be sure that you follow the instructions in the README exactly, especially
the notes about which versions of pwlib and openH323 versions work with the
current version of OH323
Regards
Scott M. Stingel
President,
Hi,
Y si te molesta no hubieras respondido OK ?
because this isn't a nufone support ML.
The next time, post your configs, not
your complains about nufone. Without that,
no one has divinatory powers and can help you.
y mas BLA BLA BLA eres TU ... porque quizas
cuando TU no habias nacido aun,
Broavoice has their problems but to tell the truth, every time i have had
a problem, i called them ans it was resolved. This is a better track
record than with the PSTN providers.
I personally use Broadvoice as incoming and an secondary outgoing. I use
voicepulse for my outgoing since they
Hi
Il giorno dom, 21-11-2004 alle 20:49 +, Mike Dent ha scritto:
Ok, so I realised I was running a CVS version of * which might have been
giving
me the SIP problems. So I decided to get down 1.0.2. I followed the usual
instructions, compiled and installed it. (FC2)
[chan_zap.so]Nov
Hi all, This may have been posted before, but it driving me crazy trying
to resolve it.
I have a new Asterisk setup, currently not much more that just 2
phones/extensions defined. But I can't get 2 SJ phones to register on the
server.
My sip.confg is unchanged with the exception of the lines
Can you actually dial the phones and is host set to dynamic? I'm new to
this too and I discovered that you only need to register when host is
set to dynamic. Which makes sense when you really think about it.
Ed
Chuck Keeter wrote:
Hi all, This may have been posted before, but it driving me
H.
Is it possible that the line is detecting a polarity event, decided that the
line is ringing and started listening for a non-existant V23 data stream, and
then the line has not in fact rung?
This would mark the line as busy (and unable to handle an outgoing call) but
when a call did in
At 04:35 PM 11/21/2004, you wrote:
Can you actually dial the phones and is host set to dynamic? I'm new to
this too and I discovered that you only need to register when host is set
to dynamic. Which makes sense when you really think about it.
Ed
Nope, I can't dial. it stops at waiting to
On Sun, Nov 21, 2004 at 04:34:38PM -0600, [EMAIL PROTECTED] wrote:
[chan_zap.so]Nov 21 20:37:05 WARNING[-150101888]: loader.c:248
The warning-number your quoted, -150101888, is actually the process
id of the thread the error gets reported about.
See also
Thank you for reply Edwin. I just managed to fix this by downloading a fresh
libpri and compiling it again.
Asterisk starts now, if only this phone would!
Mike
On Mon, 22 Nov 2004 09:46:53 +1100, Edwin Groothuis [EMAIL PROTECTED] wrote:
On Sun, Nov 21, 2004 at 04:34:38PM -0600, [EMAIL
What headsets have people found work well with the Cisco 7940 and 7960
phones? To date, I have tried a couple of the headsets within the
Plantronics H series (H41-N), and noticed that the volume of my speaking
is lower over the headset than on the regular handset. I am currently
looking for
What headsets have people found work well with the Polycom Soundpoint
500 600? If any of the headsets that work with the Polycom's also
work the Cisco, that would be an added bonus, but is by no means a
requirement. I am not looking for a headset that requires an external
amplifier, but
On Sun, 21 Nov 2004 18:01:07 -0500, Brian Pavane
[EMAIL PROTECTED] wrote:
What headsets have people found work well with the Cisco 7940 and 7960
phones? To date, I have tried a couple of the headsets within the
Plantronics H series (H41-N), and noticed that the volume of my speaking
is lower
I have used the M175, which is a convertible(over the head or the ear loop)
and has a volume control for the microphone and the earpiece and a mute
switch for the mic.
Lyle
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Bob Goddard wrote:
Not all over $500 - a quick search finds:
http://www.xenarcdirect.com/search_results.asp?txtsearchParamCat=6txtsearc
hParamType=ALLtxtsearchParamMan=ALLtxtsearchParamVen=ALLiLevel=1
Product ID: 700TSCategory: 7 LCD Monitor
700TS - 7' USB Touch Screen LCD Monitor with VGA
Hello,
I have a sattelite link (expensive link) and we are
very concern with our bandwidth usage. I would like to force asterisk to place
outgoing call(over the sattelite)to g729 codec no matter what codec comes
to asterisk.So asterisk will be making translation from the incoming codec
Exactly correct Paul.
Thank you!
Jay
- Original Message -
From: Paul Dugas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, November 20, 2004 10:11 PM
Subject: Re: [Asterisk-Users] ANY DEVELOPERS HERE?warning:implicit
On Sat, Nov 20, 2004 at 09:11:15PM -0800, Tracy R Reed said:
On Sun, Nov 21, 2004 at 12:05:27AM -0500, Gregory Junker spake thusly:
What is the size of the current line panel on her desk? I am thinking it
might be worthwhile to produce an addon to Asterisk that drives a flat
touchpanel
What VOIP Phones is everyone using and why? Is the a common phone that seems the work the best? Just wondering. --Tony
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To UNSUBSCRIBE or
Hi,
(B
(BI'm now setting up a VoIP conference room using Asterisk.
(B
(BAll the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most.
(B
(BSo, basically I think I can handle the situation
Kevin Brennan wrote:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
2GB?!? Remember, each voice channel you trunk across TDMoE is 64Kbps.
While overprovisioning is laudable,
Gregory Junker wrote:
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Bayonne is supposed to act as a key system, at least that's what was
described on the
I have an application that I want to be able to verify that the call coming
in on a PSTN 800 number is from an authorized caller.
I want to read the CallerId then terminate the call without answering it.
exten = s,1,Wair(3)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Hangup()
Any ideas
I have a Plantronics M12 amplifier and a bunch of interchangeable
headsets. I haven't found anything that this won't work on yet.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Sunday, November 21, 2004 5:45 PM
To: Shaun Ewing;
On Sun, Nov 21, 2004 at 04:25:39PM -0800, Tony Vickers spake thusly:
What VOIP Phones is everyone using and why? Is the a common phone that
seems the work the best? Just wondering.
My picks in increasing order of quality and price:
Grandstream - It's cheap and looks funny but it works.
Snom -
Put all authorized CallerID into Asterisk database (on cli: database put
allowedcaller 1234567 1) and then do a lookup, whether CallerID is
allowed. (1234567 is CallerID)
exten = s,1,SetVar(allowed=0)
exten = s,2,DBget(allowed=allowedcaller/${CALLERID})
exten = s,3,GotoIf($[${allowed}]?5)
exten
I've tried 5 different headsets, some amplified, some using the headset jack.
I've settled on one. The Toughset 10, is very comfortable, and has
excellent sound quality.
http://www.vxicorp.com/storefront/detail.asp?PRODUCT_ID=200585l2=callcenter
-Chris
On 03:01 PM 11/21/2004, Brian Pavane
On Sun, 21 Nov 2004 16:25:39 -0800 (PST), Tony Vickers
[EMAIL PROTECTED] wrote:
What VOIP Phones is everyone using and why? Is the a common phone that seems
the work the best? Just wondering.
I really like the Cisco 7960 - but its also a bit pricey.
At Astricon I was lucky enough to get a
Thanks, but that does not actually terminate the call. The phone continues
to ring until the caller hangs up.
I have done an application with cellphones that allowed allowed me to send a
signal to the phone company to drop the call. Maybe this is just a cell
phone thing.
George
Put all
On Sun, 2004-11-21 at 20:50 -0500, Leif Madsen wrote:
On Sun, 21 Nov 2004 16:25:39 -0800 (PST), Tony Vickers
[EMAIL PROTECTED] wrote:
What VOIP Phones is everyone using and why? Is the a common phone that seems
the work the best? Just wondering.
I really like the Cisco 7960 - but its
Can some people post some configurations they've implemented when
deploying an * system for let's say 25-50 stations and maybe a larger
200 station system? I would assume some kind of chassis with some DSP
boards and some kind of system board with a hard drive for running the
system and
Asterisk runs on any PC hardware that runs Linux, from that old PII
sitting in the closet gathering dust to that 4-way Xeon blade server in
a rack, and beyond, and all points in between. Digium has a line of PCI
cards that work with Asterisk for T1/E1 lines, ISDN PRIs, analog POTS
lines, etc.
On Sun, 2004-11-21 at 20:53, George Burt wrote:
Thanks, but that does not actually terminate the call. The phone continues
to ring until the caller hangs up.
I have done an application with cellphones that allowed allowed me to send a
signal to the phone company to drop the call. Maybe
On Mon, 2004-11-22 at 13:48, David Boyd wrote:
On Sun, 2004-11-21 at 20:53, George Burt wrote:
Put all authorized CallerID into Asterisk database (on cli: database put
allowedcaller 1234567 1) and then do a lookup, whether CallerID is
allowed. (1234567 is CallerID)
exten =
Chris Olson wrote:
Hello,
I have firefly installed and it is somewhat working. It is registering
with my Asterisk server and I can call out, but I receive no audio
coming into Firefly. From the Asterisk end, everything looks OK with
the call, just no audio is being received on the Firefly end.
Here is another user that doesn't seem to know how to setup spam
filtering... Someone please remove them from the list.
BTW, looks like their bayes DB needs to be re-trained :)
Regards,
Adam
-Forwarded Message-
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: Re:
On a pre 1.0 version I was running I patched queue.c to add estimated hold time
announcements.
Stable 1.0 (cvs checkout -rv1-0_stable asterisk) does not appear to of included
this patch and of course patching the current
queue.c with the patch I have fails.
I looked at the Matis to see if an
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