Menu - Display " -- rESEt --", please be very CAREFUL hereA Key in the physical / MAC address on back of thephone, Press Menu, phone will be reset back toFACTORY DEFAULT setting, all your settingwill be erased and gone.
B Press Menu without key in anything, phone willfunction the same as power
I'll be there...
Zoa.
Steven Sokol wrote:
On 9/16/05, Brian Roy [EMAIL PROTECTED] wrote:
On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote:
Hi,
I'm taking a straw-poll to see who out there is planning on going to
AstriCon.
Enjoyed it last year, but putting it on the west coast
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Damon Estep wrote:
Do you simply replace the .gsm files with .wav files and it plays them
in these apps, or is there more to it?
I am talking about the built in functionality of vm, queues, agents --
not the
Hi,
I have a small callcenter with 3 agents who login using
AgentCallbackLogin. They normally receive calls, but needs to call
outside also. When they call outside, though they are busy the show
agents shows them as available, and calls gets routed to them. How can
I make them busy when they
Have you found any information yet about this? I am looking for good and affordable phones that can use DNS myself, but not for failover, simply for ease of use by some non-computer savvy family members. So far, I am afraid I'm going to be limited to USB/software phones. I would greatly appreciate
Checkout this.
http://lists.digium.com/pipermail/asterisk-users/2005-July/116881.html
I borrowed the structure from somewhere else that I found it..
Since then I've moved primarily to postgres through odbc for myself.
-bill
On 16-Sep-05, at 2:44 PM, lenz wrote:
Thanks, is there a standard
On Sat, 17 Sep 2005, Zoa wrote:
I'll be there...
Zoa.
I'm keen to be there. Cape Town to LA - must be flying time of 20 hours
or more, so stop complaining everyone...
Steve
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I'd like to use the VoIP service from Free with Asterisk,
but am having a couple of problems. Here are some details:
ADSL from Free Télécom comes bundled with VoIP and TV
services. Most users access the VoIP via the supplied
Freebox, which is an integrated DSL modem, router, ATA, and
media
Hi. I am using the Flash operator panel 0.24 and it
works, but I don't see the voicemail icon when I have incoming
voicemail.
In the op_buttons.cfg I have the following
setup:
[SIP/100]
Position=2
Label="Office tel. 1"
Extension=100
Icon=1
Mailbox=100
I've tried to google on the subject,
Tony Mountifield wrote:
Do wav or sln versions exist of the standard Asterisk sounds by Allison?
I mean the versions before GSM compression was applied, not just ones
obtained by uncompressing the GSM again.
Unfortunately they have not been preserved :-(
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Damon Estep wrote:
Do you simply replace the .gsm files with .wav files and it plays
them
in these apps, or is there more to it?
I am talking about the built in functionality of vm, queues,
agents --
not the
Damon Estep wrote:
# for a in *.wav; do sox $a `echo $a|sed -e s/wav//`ul ; done
I'll even give you another helpful hint (assuming you are using bash):
# for a in *.wav; do sox $a ${a%.wav}.ul; done
'man bash' is interesting reading :-)
___
Hello,
On 9/17/05, Insider KT [EMAIL PROTECTED] wrote:
Hi. I am using the Flash operator panel 0.24 and it works, but I don't see
the voicemail icon when I have incoming voicemail.
In the op_buttons.cfg I have the following setup:
[SIP/100]
Position=2
Label=Office tel. 1
Damon Estep wrote:
# for a in *.wav; do sox $a `echo $a|sed -e s/wav//`ul ; done
I'll even give you another helpful hint (assuming you are using bash):
# for a in *.wav; do sox $a ${a%.wav}.ul; done
'man bash' is interesting reading :-)
Yeah, it shows, I am no linux expert...
Is
Hi,
I have a small callcenter with 3 agents who login using
AgentCallbackLogin. They normally receive calls, but needs to call
outside also. When they call outside, though they are busy the show
agents shows them as available, and calls gets routed to them. How
can
I make them busy when
On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote:
Hi,
I'm taking a straw-poll to see who out there is planning on going to
AstriCon. I would like to hear from both new members of the community
and gurus. What kinds of things would you like to see at an Asterisk
Conference? What topics
Damon Estep wrote:
Is it safe to assume that having all of the prompts in a format that
does not need to be transcoded will result in less cpu time with the
same call load?
Absolutely.
Would it also make sense to do the same for MOH and move away from mp3?
MOH with native files seems to be
Zoa wrote:
We just uploaded the latest and greatest version of the idefisk iax2
softphone, version 1.24
Freely downloadable at:
http://www.asteriskguru.com/tools/idefisk_beta.php
Changes since the last release include:
- history panel is working
- receiving messages and urls (sendtext
For your outbound calling problem, if you're operating with CVS-HEAD you can PauseQueueMember and then UnpauseQueueMember as part of the dial-plan for your outbound calls for those agents.
On 9/17/05, Rajkumar S [EMAIL PROTECTED] wrote:
Hi,I have a small callcenter with 3 agents who login
[snip]
As for topics, I enjoyed Madrid, but feel there are 2 tracks to be
seen there:
1) pragmatists - What can I do with Asterisk? Business Drivers.
For them you want case studies, some forward looking announcements, and
loads of demos of edgy configs. Plus economics - eg 8 line PRI
BJ Weschke wrote:
For your outbound calling problem, if you're operating with CVS-HEAD
you can PauseQueueMember and then UnpauseQueueMember as part of the
dial-plan for your outbound calls for those agents.
Thanks, I think this will do the trick. For short breaks, I can wrap
this around an
Steven Sokol wrote:
That is an awesome suggestion! We'll do it! We have a room we've
labeled the Email Garden. We'll rename it the Code Domain or
something and try to get at least one guru to man the desk in there,
dispensing advice as well as pizza and Red Bull. The room was already
set to
It might be nice to have con for each coast? Maybe each region?
(Boston, Atlanta, and uhtwo large cities out west :-)) It's a lot
of work ,but you might get a better turn out because people won't have
to travel to Atlanta or someplace like that
Just my .02.
~kurth
Does anybody has intalled it on freebsd with unixodbc or libiodbc and have
it working?
Regards,
Olivier
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On Sep 17, 2005, at 7:08, Steven Sokol wrote:
So far we've had at least one person indicate that they would not want
to travel to the US at this time. All politics aside, how many out
there feel the same way? I do NOT want to start a polical flame-war,
but I am curious at the number of people
On 9/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sat, 17 Sep 2005, Steven Sokol wrote:
(Anybody out there want to volunteer to bring in the hardware?)
I'll bring some Digium hardware, Sirrix boards. Assuming there'll be some
security against them disappearing...
Cool!
Subject: Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)
Joseph wrote:
Why do you need to compile it?
Isn't it available as an rpm package?
I will assume he knows why he needs to compile it.
See if the source for the rpm, deb, or whatever from the distro you
are
running will build for
Olivier Taylor wrote:
Does anybody has intalled it on freebsd with unixodbc or libiodbc and have
it working?
Regards,
Olivier
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So far we've had at least one person indicate that they would not want
to travel to the US at this time. All politics aside, how many out
there feel the same way? I do NOT want to start a polical flame-war,
but I am curious at the number of people who simply won't come to the
States due to
Can you automatically call extensions? If so, the immediate workaround
would be to define extensions that set caller id and then place the
call. You could probably write a macro to do the extension definition if
you have more than a few.
Jim Gottlieb wrote:
Hi. I'm using
On 9/17/05, Kurth Bemis [EMAIL PROTECTED] wrote:
It might be nice to have con for each coast? Maybe each region?
(Boston, Atlanta, and uhtwo large cities out west :-)) It's a lot
of work ,but you might get a better turn out because people won't have
to travel to Atlanta or someplace like
Damon Estep [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Do wav or sln versions exist of the standard Asterisk sounds by
Allison?
I mean the versions before GSM compression was applied, not just ones
obtained by uncompressing the GSM again.
Not sure about that Tony, we recorded a
On 9/17/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Steven Sokol wrote:
That is an awesome suggestion! We'll do it! We have a room we've
labeled the Email Garden. We'll rename it the Code Domain or
something and try to get at least one guru to man the desk in there,
dispensing
The best place for Astri Con 2006 would definatly be
Omaha, Nebraska! ;) very central
...ah one could hope.
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
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--Bandwidth
On 9/17/05, Kenny Kant [EMAIL PROTECTED] wrote:
The best place for Astri Con 2006 would definatly be
Omaha, Nebraska! ;) very central
...ah one could hope.
Actually, we're based out of Kansas City, Missouri (NOT Kansas - we
believe in science) so Omaha would be pretty convenient for us
Hi. I
am using the Flash operator panel 0.24 and it works, but I don't see
the voicemail icon when I have incoming voicemail.
In the op_buttons.cfg I have the following setup:
[SIP/100] Position=2
Label="Office tel. 1" Extension=100
Icon=1 Mailbox=100 I've tried
to google on
Actually, we're based out of Kansas City, Missouri (NOT Kansas - we
believe in science) so Omaha would be pretty convenient for us --
it's the other 98% of the community which would have four connections.
;-)
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is
Matthew Simpson wrote:
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is hub for ATA
All good central locations with cheap non stop flights.
Atlanta is central for who? With all of the tornados, hurricanes, etc.
I would definately vote no for there. Dallas and
I currently not use it due to some limitations in * realtime .
Such as?
-Matthew
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Matthew Simpson wrote:
So far we've had at least one person indicate that they would not want
to travel to the US at this time. All politics aside, how many out
there feel the same way? I do NOT want to start a polical flame-war,
but I am curious at the number of people who simply won't come
Hello All,
I tried to implement capiFax to receive on an eicon diva server, and if
I call the msn, and hang up, the capifax starts but segfaults asterisk.
Also the system lags for about 5 seconds.
Anyone know how I can trace down the issue?
I am running head with chan_capi_cm
Regards,
Greg
Sorry I got it, needed to recompile a clean chan_capi-cm.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Saturday, September 17, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Damon, thanks for the tips on recording (below). I was particularly
interested in the 10% time reduction. I had a few questions about
it:
a) are you doing a straight speed-up, so that the frequency
increases
by 11% as well as the speed? Or are you doing a pitch-preserving
duration
On Thu, 15 Sep 2005, [ISO-8859-1] Stéphane LAVRI wrote:
Hi
I'm looking for a company who can provide me an Internet connection
between africa and Europe.
Plesa If someone can give me some contact name or company dont
hesitate to send me a mail at [EMAIL PROTECTED]
Africa is a big
Are you running it with AGI or with DEADAGI?
Darren Wiebe
[EMAIL PROTECTED]
Michael K. Rodriguez wrote:
I have been testing the ASTCC and have notice that when the caller hangs up
the line while the balance is being played back the sub savedata() is not
being called because the asterisk
Somewhere with a little more to do other than the conference for those of us
who travel from further a field or those who want to make a holiday, sorry
vacation out of the trip. I must admit, if it was held at Atlanta again I
would have to think twice about going. CNN centre, Coca Cola museum and
Trying to get moh (mp3) not to randomize playback of files, cant figure
out what order it picks. Does not appear to be alphabetical.
I do not have the r option in the MOH class, but the files are played
in an order I can figure out, they do not appear to be random either,
same pattern repeats.
I am running Asterisk 1.0.9 and the latest bristuff in combination with a
HFC isdn card, connected to a BRI interface.
For some reason, I am not able to have it dial out (see below). It exits
with DIALSTATUS=CHANUNAVAIL.
One thing that may be misconfigured is that it says: Signalling Type: PRI
On Sat, 17 Sep 2005, Stewart Nelson wrote:
I'd like to use the VoIP service from Free with Asterisk,
but am having a couple of problems. Here are some details:
ADSL from Free Télécom comes bundled with VoIP and TV
services. Most users access the VoIP via the supplied
Freebox, which is an
On 9/17/05, asterisk [EMAIL PROTECTED] wrote:
[snip]
What about holding it a week before or after one of the other voip
conventions in the same location. This might help to attract more people
who are in town early or staying on after the other conference and make
the trip more productive
Somewhat off topic, who makes the hardware that Free is using? Is it
available/being used in the US?
http://www.broadcom.com/
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Asterisk 1.09 works well with it however CVS head has major
issues, wondering if anybody else has seen any RTP like issues and what they
did to work around it.
Thanks
Chris
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Is the a way to check the voice mail of a different extension?
I know if I setup in sip.conf context name not associated with the
extension; so every time I login to check a mail it will asking me for
mail box number (extension). But if the my extension matches the
mailbox it is just asking me
Grandstream supports DNS SRV
Justin Richards wrote:
Have you found any information yet about this? I am looking for good
and affordable phones that can use DNS myself, but not for failover,
simply for ease of use by some non-computer savvy family members. So
far, I am afraid I'm going to be
Damon Estep wrote:
I do not have the r option in the MOH class, but the files are played
in an order I can figure out, they do not appear to be random either,
same pattern repeats.
Oh come on, its obvious :-)
Have you figured it out yet?
Yet?
Now?
OK... I'll tell you.
See the order you
Kevin,
That is what I thought I read somewhere, but it is not so. I will check,
but I THINK that * reads the file names left to right top to bottom and
my FC4 box lists them with an ls top to bottom left to right!
Does * actually collect the file names with an ls? Could this be a small
bug?
I
I have a Cisco 7940 phone with a locked SIP configuration. There is no
tftp server configured in network settings. Does anyone know how to get this
phone to upgrade its firmware and thereby unlock the SIP settings? (The **#
combination on the Cisco manuals does not work).
-brian
If you're wanting some of the Asian users how about somewhere in SE Asia
such as Singapore for an Astricon? Is also good for us Australians. I went
to Madrid this year for Astricon but I'm not sure I'd ever be able to make
it to the US. Besides, Singapore is only 5 hours flight from
On 9/17/05, Ed Greenberg [EMAIL PROTECTED] wrote:
I'm not going because it falls on Yom Kuppur. What's up with that?
Ed,
I am really very sorry about that. I know it will prevent some people
from making it. Unfortunately, Pulver moved VON to the spot we had
last year (this next week) and
Is there anyone else out thee that sees a need for an ackcall=yes in
queues.conf instead of just agents.conf?
I want to have a queue that is hardcoded to dial certain channels
instead of agents.
Some of the channels are cell phones that can occasionally go straight
to VM (as in phone powered
Set up an extension that goes to voicemailmain()
Exten = ,1,voicemailmain()
Dial (or whatever you want
Another option is to set it up so you call your voicemail and press *
during the greeting, this is typical of many cell carriers, add the
following to your extension macro if you use
I'd have to second what Craig mentioned. Begin based out of Singapore
we brought up a couple of points for consideration on organising an
AstriconAsia in an email to Olle sometime back. SE Asia (and
generally Asia) as a whole is really seeing a large increase in the
number of IP Service Providers
First try this:
grep -ir voicemailmain /etc/asterisk
to see if it already exists. The sample config files I started with had
extension 8500 for that and I left it there for the time being.
Damon Estep wrote:
Set up an extension that goes to voicemailmain()
Exten = ,1,voicemailmain()
Hoho，
just two days, I had dothis change to asterisk-1.0.9.
Update the queue log to mysql. It seems works well.
lenz [EMAIL PROTECTED] 写道：
Thanks, is there a standard schema for queue_log or can I define it myself?Thanksl.In data Fri, 16 Sep 2005 18:48:13 +0200, William Lloyd <[EMAIL
try going into settings and then unlock config and try typing in cisco
On 9/17/05, Brian [EMAIL PROTECTED] wrote:
I have a Cisco 7940 phone with a locked SIP configuration. There is no tftp
server configured in network settings. Does anyone know how to get this
phone to upgrade
I'm new to asterisk and need some help with ideas to handle this
configuration question.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Bockman
Sent: Saturday, September 17, 2005 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
Matthew
Well I'm stunned no one has suggested a webcast option.
I mean we aren't talking a bunch of people unable to grasp the concepts
of chat/voice/vision sessions with a log in/remote display capability.
If you think this is an option let me know I have someone who has some
software they wouldn't
On 9/17/05, Reid Forrest [EMAIL PROTECTED] wrote:
Matthew Simpson wrote:
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is hub for ATA
All good central locations with cheap non stop flights.
Atlanta is central for who? With all of the
Sounds like a great idea to me --I know of some software which would
allow audio and shared web pages and even talkback using very little
resources. Even a dialup user could use it.
http://www.talkingcommunities.com .
on Saturday 09/17/2005 Dean Collins([EMAIL PROTECTED]) wrote
Well I'm
This would be super-fantastical!!!
With all of the other conferences going on, I can only get away so much. I
love the idea of a webcast...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Saturday, September 17, 2005 8:37 PM
To:
Matt Riddell wrote:
Any other suggestions please??!
Send me a copy of a wave file with the recorded beeps after hangup and I will
see if the tones are somehow different.
Matt - did you get that wav file? Any thoughts on that one?
I have reconpiled onto debian sarge now (the main distro i
I noticed while reading some posts that people were looking for a complete
NPA-NXX list
for all area codes and prefixes.
We happen to have the entire database. So I am making it available to the
public.
Help is available at:
http://download.sixtel.net/npa/help.txt
(Caution, 20meg files)
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