By design, it seems that when an agent takes a call, the next
call is not (also) sent to his phone (strategy=ringall)
Is there a way to force asterisk to also send the next queue
call to the same agent that is on the phone? (So the agent
can answer via call waiting.)
We are using queues as a rin
[2000]
canreinvite=no
type=friend
username=2000
secret=[mypassword]
host=dynamic
context=office2
Some time, as now, the phone wroks fine, but some time is show the warning and
i cnanot use the phone...
thanks.
Michele
- Messaggio originale -Da: Gregory Junker <[EMAIL PROTECTED]>A:
As
Hello Everybody:
I am trying to know how much internet bandwidth is needed to handle twenty simultanous SIP calls.
I appreciate any help
Do you Yahoo!?
The all-new My Yahoo! Get yours free!
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Ok... so I set up a Queue and Agents.
Logging into the queue via a cisco 7960 (SIP) and
an IAXy. Both log in just fine, but the IAXy never
gets calls.
queues.conf--
[comp-noc]
strategy = ringall
member => SIP/6234445657
member => Agent/@2 ; Any agent in group 2
agents.conf---
What has always amused me is how finely folks will split hairs when they
are doing something wrong.
Brian West wrote:
No guys call it what it is. Copyright Infringement.
You can't have it both ways boys.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PRO
http://www.cdw.com/shop/products/default.aspx?EDC=568864
Can anyone tell me if this switch will be able to supply a Cisco 7940
phone with power? I've heard of PoE issues with differing switches and
the like, and I don't know how to check to see if this switch will be
able to drive the phone in
On 17/11/2004 16:13 Lex Lethol said the following:
Is there a list of loadzone/defaultzone country codes included in
zonedata.c? Can this list be updated to include my country? (I have
the tones)
edit zonedata.c, add in your tones and recompile/reinstall the zaptel
drivers. also, since libtonezone
On 18/11/2004 04:17 Steven Critchfield said the following:
On Wed, 2004-11-17 at 11:49 -0800, Tracy R Reed wrote:
And it seems to be something the developers are not interested in
supporting. Whenever someone asks about this feature they are normally
told that this is a feature of small-office "key
On Thu, 18 Nov 2004, Daniel wrote:
> >On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
> >You *can* play a welcome message without >answering the line, however,
> >this doesn't always work. eg, I tried this >config on my PRI in Australia
> >(Telstra) and:
> >
> >a) Calling from a standard analog lin
In my house i am using an autodialer to dial 74949000 to access to
gateway and then i dial my mobile or local number to benefit from the saving
Can we do that in asterisk to autodial to the gateway 74949000 and wait
10 before i eneter my destination number
Please advise me on that how to make a
Hi,
Voicemoil capabilities are in Asterisk. You can use Asterisk
voicemail from any SIP Software.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ashling
O'Driscoll
Enviado el: miércoles, 17 de noviembre de 2004 18:28
Para: [EMA
Peter Childs wrote:
http://sunflowerhead.com/software/yac/index.html
You only need to run the client..
much nicer solution then the winpopup suggestion, I've actually been
looking for some tray bubble app like this for a while that was a mini
sip client, guess this will do in the mean time...
isk below 80 G, RAM below 2x128 M
> >>anyway.
> >>
> >>What is the recommendation for the the power?
> >>
> >>bye
> >>
> >>Ronald
> >>___
> >>Asterisk-Users mailing li
Hi all, I am very intrested in that question too. Anyone pls can share
his experience with CCM in 2 words?
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You only need to run the client..
exten => s,4,System(/bin/echo -e "'@CALL${CALLERIDNAME} ~${CALLERIDNUM}'" |
nc -q 0 -w 1 pjcm400 10629 )
Does the trick for me... and YAC has a nice caller history log etc (and
I do like those nice windows b
No guys call it what it is. Copyright Infringement.
You can't have it both ways boys.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Sean Kennedy
> Sent: Wednesday, November 17, 2004 11:01 PM
> To: Asterisk Users Mailing List
Anyone know how return codes work and how I can use them in
my dialplan?
I am trying to get my system to monitor how many agents are
logged into a queue. When the queue is empty the system will
forward the call to an outside number.
I tried setting a globalvar to the total number of agents
logged
Eric Wieling wrote:
Kyle Hagan wrote:
Alot of people have needed the SIP firmware for the Cisco 79xx phone.
I found a link for them..
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html
Well, yeah, I guess stealing it is one way.
I wouldn't ca
Has Anybody got asterisk working with Diva 4bri and fedora core 2?
No matter what I try I can not get chan_capi to compile.
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> Actually, this is required to work for telco's (I would think this is
> the same in most countries). Consider premium rate phone services (in
> Australia, 1-900 xxx xxx) where you are charged $x per 'time unit'. eg,
> $5/minute etc... The service operator is required to tell you how much
> the c
I thought I might try this, but smbclient returns an error - cannot resolve
host - a PING of the host is fine.
Any ideas?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Hutton
Sent: Thursday, 18 November 2004 15:24
To: AsteriskUserMaillist
Hello everybody
I wonder if I can get any good references of web-based management
interfaces of Asterisk?
I would rather prefer more on detailed functionalities that a great
look, I am willling to work on provide a UI support in case there's
one in progress
Also, is there's some for a price I wo
On Wed, 17 Nov 2004, Noah Miller wrote:
> I'm ordering some more phones - I have the Polycom IP 500's now and I
> like them. I need some less expensive phones, and I'd like to stay
> with all Polycoms for ease of administration. I've heard, though, that
> the IP 300's don't support PoE even t
Here's a tested example that works without any scratch file. I still
had to use a combination of single and double quote characters, as well
as a double backslash for the \n newline command.
; Extension 200 Call ID Popup Example
exten => 200,1,NoOp(${CALLERID} ${DATETIME})
exten => 200,2,System(/
Kyle Hagan wrote:
Alot of people have needed the SIP firmware for the Cisco 79xx phone. I
found a link for them..
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html
Well, yeah, I guess stealing it is one way.
__
Tim Courcy wrote:
That's not true the IP600 supports POE on the phone with standard cat-5
cable.
Ahh, glad to be corrected. Thanks.
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Hi
Recently the workgroup in which I'm working buy a E100P card, and we are interested to build a Media Gateway controlled by MGCP protocol. It's possible build that using asterisk operating in Media Gateway Mode??
The information that i have say that the asterisk can work only in call
Just wondering if anyone has used either of these motherboard with a
TE405p. My current board is causing problems, and I'm looking to replace
it...
gigabyte GA-7NF-RZ
gigabyte GA-7N400 Pro2
Thanks,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 8304 [EMAIL P
Thomas Hutton wrote:
Hi Duane,
You asked "Why dump to a file?" - I don't know if this is possible or
not, but can you send a d to the smbclient -M command? I believe
the way you wrote the command it will just hang, no?
the equiv of ctrl+d is hit when it runs out of things to echo to the
process..
Hi Duane,
You asked "Why dump to a file?" - I don't know if this is possible or
not, but can you send a d to the smbclient -M command? I believe
the way you wrote the command it will just hang, no?
Thomas Hutton
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Adam Goryachev wrote:
Also, you can use something like (untested code):
exten => 200,2,System(/bin/echo -e 'Incoming Call From: ${CALLERID}'\n
Received: ${DATETIME}\n > /tmp/asterisk/${UNIQUEID}
exten => 200,3,System(/usr/bin/smbclient -M target_netbios_name <
/tmp/asterisk/${UNIQUEID})
why dump to
When I call via * my bank and an automated system ask me to enter the
numbers they aren't recognized by the system. I'm getting an error
message that the numbers I entered are invalid. WHY?
--
#Joseph
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On Thu, 2004-11-18 at 13:51, Thomas Hutton wrote:
> Hey, thanks to everybody who posted to my earlier thread. Here's a
> solution I came up with based on reading your scripts and advice.
>
> It's really simple and stupid- but seems to work great. Incoming
> calls for any type of extension can be
Hey, thanks to everybody who posted to my earlier thread. Here's a solution I came up with based on reading your scripts and advice.
It's really simple and stupid- but seems to work great. Incoming calls for any type of extension can be configured to make winpopups (or linpopups : ) on any l
That's not true the IP600 supports POE on the phone with standard cat-5
cable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, November 17, 2004 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Thu, 2004-11-18 at 11:28, Daniel wrote:
> >On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
> >> On Nov 17, 2004, at 1:30 PM, Eric Wieling wrote:
> >You *can* play a welcome message without >answering the line, however,
> >this doesn't always work. eg, I tried this >config on my PRI in Australia
>
I did not know what enum was, or where it was. I just knew that IAXtel
has been down most of the time that I have tried them.
Thank you for the update, but my offer still stands open.
If any one is interested, just email [EMAIL PROTECTED]
I in no way am trying to compeate with, or replace ENUm o
>On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
>> On Nov 17, 2004, at 1:30 PM, Eric Wieling wrote:
>>
>> > Joe Greco wrote:
>> >>> Who to generate "ring tone" to a calling party when the call is
>> >>> passed
>> >>> to an extension.
>> >>>
>> >>> The asterisk answers correctly, plays welcome me
Christopher Dobbs wrote:
What ever, Just trying to be a help to the system.
The benefit of enum over a relay service is the ability to interoperate
with others using SER and other VoIP PABXs as well, rather then being
limited to just other asterisk users, it's self managed via the web
interface
What ever, Just trying to be a help to the system.
Duane wrote:
Christopher Dobbs wrote:
Let me clerify, Send a username and password for use on my new IAX
relay system.
It wont use real phone numbers, but it will work to link the /free
world/ of IAX users.
Why not just use www.e164.org via enum
Christopher Dobbs wrote:
Let me clerify, Send a username and password for use on my new IAX relay
system.
It wont use real phone numbers, but it will work to link the /free
world/ of IAX users.
Why not just use www.e164.org via enum lookups then, it does let you use
both real phone numbers (afte
Send account request information to [EMAIL PROTECTED]
Christopher Dobbs wrote:
Let me clerify, Send a username and password for use on my new IAX
relay system.
It wont use real phone numbers, but it will work to link the /free
world/ of IAX users.
___
Let me clerify, Send a username and password for use on my new IAX relay
system.
It wont use real phone numbers, but it will work to link the /free
world/ of IAX users.
Kevin Walsh wrote:
Christopher Dobbs [EMAIL PROTECTED] wrote:
Given that IAXtel has not been responding for some time, I am
Adam Goryachev wrote:
On Thu, 2004-11-18 at 05:43, Leandro Morgado wrote:
Matt Riddell wrote:
Leandro Morgado wrote:
Although, I still think that there is some kind of incompatibility or
battery drop timing problem between Asterisk and the Nokia 32. I wish
I knew more about telecom
Christopher Dobbs [EMAIL PROTECTED] wrote:
> Given that IAXtel has not been responding for some time, I am willing to
> setup accounts for thoes who want to have that kind of functionallity.
> If you are interested, send me an email with your requested username and
> password, and i will send you y
Matt Riddell [EMAIL PROTECTED] wrote:
> Stephen R. Besch wrote:
> > This all reminds me so much of Jonathan Swifts bit about the BigEndians
> > and the LittleEndians (referring to which is the 'correct' end to open a
> > soft boiled egg) in Gulliver's travels.
> >
> But that's simple, surely you sh
Given that IAXtel has not been responding for some time, I am willing to
setup accounts for thoes who want to have that kind of functionallity.
If you are interested, send me an email with your requested username and
password, and i will send you your account information.
_
On Thu, 2004-11-18 at 12:05, Chad Scott wrote:
> On Nov 17, 2004, at 1:30 PM, Eric Wieling wrote:
>
> > Joe Greco wrote:
> >>> Who to generate "ring tone" to a calling party when the call is
> >>> passed
> >>> to an extension.
> >>>
> >>> The asterisk answers correctly, plays welcome message and
On Thu, 2004-11-18 at 05:43, Leandro Morgado wrote:
> Matt Riddell wrote:
>
> > Leandro Morgado wrote:
> >
> >> Although, I still think that there is some kind of incompatibility or
> >> battery drop timing problem between Asterisk and the Nokia 32. I wish
> >> I knew more about telecomms and wc
[snip]
> > I got it:
> > Dial(channel, timeout, r)
> >
>
> If the destination is busy you may still hear ringing.
When the phone is busy it jumps straight to next next line which is
"voicemail"
--
#Joseph
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I thought Asterisk would indicate ringing on the PRI if it hadn't
answered it yet. If you answer the PRI and then Dial() don't you get
silence?
Since he says he's played a welcome message, I'd think the line has
been answered and therefore he has to indicate ringing to the dialing
party with
I think maybe some of them come from a telco world and not the real world?
:)
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Off topic, but this seems like a good place to ask :)
Why do vendors think it is in their best interest to encrypt device config
files and then restrict access to the tools to create the config files?
If I made a product, I'd be "bending over backwards" to make sure
everybody who wanted my produ
Joe Greco wrote:
I have to confess that I don't understand the "real problem", then.
What else are you supposed to do when you've already answered a channel
and you then want the caller to hear ringing? It didn't seem to work
automatically, and I'm certainly not convinced that it should.
That depe
Joseph wrote:
On Wed, 2004-11-17 at 14:58 -0700, Joseph wrote:
[snip]
Asterisk will ALWAYS indicate ringing if it can.
The "r" option to Dial, Playtones, and Ringing are all hacks/workarounds
for when Asterisk cannot indicate ringing to the calling party. You
should diagnose and fix the real pro
Andrew Kohlsmith wrote:
On November 17, 2004 04:30 pm, Eric Wieling wrote:
Asterisk will ALWAYS indicate ringing if it can.
The "r" option to Dial, Playtones, and Ringing are all hacks/workarounds
for when Asterisk cannot indicate ringing to the calling party. You
should diagnose and fix the real
Hi all,
what is the meaning of this message:
Nov 17 19:18:27 NOTICE[514032]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'us'
Regards
Bastian
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http:
Alot of people have needed the SIP firmware for the Cisco 79xx phone. I
found a link for them..
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html
Kyle
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for solaris 10 x86, can we use digium hardware? any success?
On Wed, 17 Nov 2004 19:23:16 +, Marty Lee
<[EMAIL PROTECTED]> wrote:
>
> > Bob Knight wrote:
> >
> >
> >> I am waiting for solaris 10 for x86.
> >
> >
> > You can download 32 bit versions now.
> > I just downloaded the sparc vers
On November 17, 2004 05:21 pm, Adam Hart wrote:
> try running ethereal, make sure everything looks ok and send me the
> result. No firewall? Also, download debugview from www.sysinternals.com
> to see Firefly's debug msgs. Could be simply wrong audio device?
No, I hear the odd ringback tone in my
Stephen R. Besch wrote:
This all reminds me so much of Jonathan Swifts bit about the BigEndians
and the LittleEndians (referring to which is the 'correct' end to open a
soft boiled egg) in Gulliver's travels.
But that's simple, surely you should put the big end of the egg into the
egg cup and op
On Wed, Nov 17, 2004 at 05:07:33PM -0500, Bob Willock spake thusly:
> I just bought a couple of these Cisco 7970G phones and it seems that they
> require a SIP image binary file to load when the phone boots and this file
Cisco phones are hugely overrated. I have deployed a number of them and I
hav
Noah Miller wrote:
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay with
all Polycoms for ease of administration. I've heard, though, that the
IP 300's don't support PoE even though their brochures say they
The phone has to be registered with Asterisk first. What is your setup
in sip.conf for this phone?
Greg
Michele wrote:
Hello, this is my first message on thi ML;
I'hava a problem: I have a voismart 101 phone and at the moment of registration
or when I make a call,in the asterisk's consolle i can
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though, that
the IP 300's don't support PoE even though their brochures say they do.
Has anybody
Hi Frank. When I try to compile using the patched fiels I get the
following error
[EMAIL PROTECTED] capi]# make
gcc-2.95 -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/usr/asterisk/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6
-DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-
On Wed, 2004-11-17 at 16:17 -0600, Shaun Tierney wrote:
> When I use the Dial command to connect a call using my Asterisk PBX, it
> seems that the PBX says that the call was answered right when the two
> channels are bridged together, rather than when the actually callee answers
> their phone. I w
HEAD 20041117 (also tried with 20040806 and
200410-something)
IAX2, no NAT. Firefly->Asterisk audio works, but I can't hear anything from
the other side.
Using GSM codec, also tried ulaw.
Any ideas?
-A.
relevant bits of iax.conf:
[andrew-bt]
type=peer
host=dynamic
trunk=no
[andrew-
Hello, this is my first message on thi ML;
I'hava a problem: I have a voismart 101 phone and at the moment of registration
or when I make a call,in the asterisk's consolle i can read thi warning:
Nov 17 23:09:50 WARNING[2806]: chan_sip.c:683 retrans_pkt: Maximum retries
exceeded on call [EMAIL
sorry wasn't paying too close attention too the model number, the other
current reply addresses that.
Just pay the $7 or $10 for the firmware license already, sheesh.
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When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual call status and
respon
On Wed, 2004-11-17 at 22:34 +0100, Håkan Persson wrote:
Hi!
I just bought a Cisco 7960G and I want to convert it into a SIP phone.
All information to do this seems to be at
http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml
but the firmware download link o
*puts on flame suit*
(bah ignore the other prematurely-sent reply)
Seriously, though. It's not a scam, it's their business model (which is
shared by many many companies). The software license is separate from
the hardware. Always has been. You probably should have known that
before buying the p
Bob Willock wrote:
I just bought a couple of these Cisco 7970G phones and it seems that they
require a SIP image binary file to load when the phone boots and this file
updates the firmware of the phone to run in SIP mode. The only problem is
that Cisco seems to want to profit from the phone sales a
*puts on flame suit*
Bob Willock wrote:
I just bought a couple of these Cisco 7970G phones and it seems that they
require a SIP image binary file to load when the phone boots and this file
updates the firmware of the phone to run in SIP mode. The only problem is
that Cisco seems to want to profit f
I just bought a couple of these Cisco 7970G phones and it seems that they
require a SIP image binary file to load when the phone boots and this file
updates the firmware of the phone to run in SIP mode. The only problem is
that Cisco seems to want to profit from the phone sales and then block you
f
On Wed, 2004-11-17 at 14:58 -0700, Joseph wrote:
> [snip]
> > Asterisk will ALWAYS indicate ringing if it can.
> >
> > The "r" option to Dial, Playtones, and Ringing are all hacks/workarounds
> > for when Asterisk cannot indicate ringing to the calling party. You
> > should diagnose and fix the
> Joe Greco wrote:
> >>Who to generate "ring tone" to a calling party when the call is passed
> >>to an extension.
> >>
> >>The asterisk answers correctly, plays welcome message and ring an
> >>extension, but the caller does not here the rings.
> >
> >
> > Did you tell Asterisk to indicate ringin
On November 17, 2004 04:30 pm, Eric Wieling wrote:
> Asterisk will ALWAYS indicate ringing if it can.
> The "r" option to Dial, Playtones, and Ringing are all hacks/workarounds
> for when Asterisk cannot indicate ringing to the calling party. You
> should diagnose and fix the real problem, rather
[snip]
> Asterisk will ALWAYS indicate ringing if it can.
>
> The "r" option to Dial, Playtones, and Ringing are all hacks/workarounds
> for when Asterisk cannot indicate ringing to the calling party. You
> should diagnose and fix the real problem, rather than try hiding the issue.
How do you
>> Are you sure you are doing the right sequence of buttons. If you
>> have a call coming in you must flash then * 0. the flash button
>> flashes the PBX and * 0 flashes the CO.
> Yes just tried it again. Using the fxo I dialled a sip phone on the
> network, then I tried to send a flash and
We have been using g729 for over 2 weeks now. Not a single sound file is
stored in g729. Everything is WAV.
The only time the g729 is used is when someone accesses their voicemail.
HOWEVER, for some unknown reason, this came up today:
Nov 17 15:39:13 WARNING[1125327680]: file.c:779 ast_streamfile
Matt Riddell wrote:
Leandro Morgado wrote:
Correction! I do need busydetect (i had forgotten to comment it out) to
detect hangups. I'm not as familiar with this Smartcell GSM terminal but
I dont think it drops the battery when call are hung up (in
/var/log/messages voltage stayed at 9V during the c
Hi!
I just bought a Cisco 7960G and I want to convert it into a SIP phone.
All information to do this seems to be at
http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml
but the firmware download link on the page takes me to a page that
informs me that "There are c
> From: Steven Critchfield <[EMAIL PROTECTED]>
> On Wed, 2004-11-17 at 11:49 -0800, Tracy R Reed wrote:
>> On Wed, Nov 17, 2004 at 01:13:57PM -0500, Noah Miller spake thusly:
On our current phones (Iwatsu) we have a button on the phones for
each extension that lights up when that
>>>
>>
On Wed, 2004-11-17 at 15:25 -0600, Matthew Boehm wrote:
> OK. That works. Changed it to 2 seconds. Now it says:
>
> pbx_extension_helper: No application 'rxfax' for extension (all-incomming,
> fax, 2)
>
> Did I miss something? I don't see app_rxfax anywhere
rxfax has to be downloaded from St
On Wed, 2004-11-17 at 16:03 -0500, Andrew Kohlsmith wrote:
Perhaps a better idea is to have extension appearances (similar to a CAP
module) and a number of parking slot appearances. This, IMO, is far more
worthy of development work, as it scales much better. Call pickup still
works from
On Wed, 2004-11-17 at 15:23 -0600, Joe Greco wrote:
> > How to generate "ring tone" to a calling party when the call is passed
> > to an extension.
> >
> > The asterisk answers correctly, plays welcome message and ring an
> > extension, but the caller does not here the rings.
>
> Did you tell Ast
Joe Greco wrote:
Who to generate "ring tone" to a calling party when the call is passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
Did you tell Asterisk to indicate ringing?
Asterisk will ALWAYS indicate rin
On Wed, 17 Nov 2004 17:05:55 -0300, Thomas Hutton <[EMAIL PROTECTED]> wrote:
> Question: Does anyone know of a lightweight popup method to put an
> incoming call ID string on a client machine? Something as simple as
> winpopup would work great- for example: I have a call coming in on Zap/4
> but t
OK. That works. Changed it to 2 seconds. Now it says:
pbx_extension_helper: No application 'rxfax' for extension (all-incomming,
fax, 2)
Did I miss something? I don't see app_rxfax anywhere
Matthew
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Use
On Wed, 17 Nov 2004, Joe Greco wrote:
> > I don't think this is really a key system. AFAIK a traditional key system
> > has a one-to-one mapping between lines and the buttons. Some pbx:es offer
> > a mode where each *extension* is / can be represented by a button. This is
> > called a Busy Ligh
> Who to generate "ring tone" to a calling party when the call is passed
> to an extension.
>
> The asterisk answers correctly, plays welcome message and ring an
> extension, but the caller does not here the rings.
Did you tell Asterisk to indicate ringing?
... JG
--
Joe Greco - sol.net Network
No that message is normal.. I even see it when stuff is configured correctly
and working. :P
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Henry Devito
> Sent: Wednesday, November 17, 2004 3:10 PM
> To: 'spectro'; 'Asterisk Use
Who to generate "ring tone" to a calling party when the call is passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
--
#Joseph
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Asterisk-Users mailing list
[EMA
> On Wed, 17 Nov 2004, Jason Becker wrote:
>
> > > On our current phones (Iwatsu) we have a button on the
> > > phones for each extension that lights up when that
> > > extension is ringing or is in a call, so I can see at
> > > a glance if one of my coworkers is on the phone before
> > > I go bar
>When I'm using my telephone connected to the FXO I can see/hear on my
handset that call waiting is enabled, however when I attempt to send a
hook
flash or use the combination "*0" I don't get a dialtone and the feature
doesn't work. The same goes for three way calling and cancallforward.
Are you
Here's a thought, Just to be sure you have the cables and everything pinned
out correctly. Set the digital trunk card to a standard T1 in the Legend
and set * as a standard T1 and try just plain tie lines between them. If
this works then you know cards & cables are ok. As far as the message can
n
On Wed, 2004-11-17 at 14:51 -0600, Kristian Kielhofner wrote:
> Jim Dossey wrote:
>
> > On Wed, 2004-11-17 at 13:18 -0600, Kristian Kielhofner wrote:
> >
> >>Hello everyone,
> >>
> >>Since releasing my very beta, test version of AstLinux almost two weeks
> >>ago, there have been over 300 dow
On November 17, 2004 02:49 pm, Tracy R Reed wrote:
> And it seems to be something the developers are not interested in
> supporting. Whenever someone asks about this feature they are normally
> told that this is a feature of small-office "key" systems and that
> Asterisk has its sights set on bigge
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