Hello,
on my ISDN phone I can configure that on the next outgoing call, my
telephone number should not be transmitted, instead it should be UNKNOWN.
How can I configure Asterisk to do the same? Is this a feature/parameter
of the driver (chan_capi) that I'm using?
BTW: I'm using ISDN and
On PRI SetCallingPres works fine it should work with ISDN because its
the same signaling.
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set CallerID Presentation
[Description]
SetCallerPres(presentation): Set Caller*ID presentation on a call.
Valid presentations are:
As people have sugested the ATX power supplies can work without a mobo
One thing to watch out for your setup is the actual ampere requirments for
your disks
i.e Your power supply provides 300W but this is partitioned to different
voltages (+5, +12, etc) with different amp charecteristics
Disks
On Tuesday 13 May 2008, Steve Totaro wrote:
Can you describe exactly how you are utilizing it, including LAN/WAN,
switches, ping times, and other network central details. TDMoE adds
the E (ethernet) component to troubleshooting and I think do to this,
it may be very fragile depending on
Thanks. If I find out some settings for soxmix, do you maybe know where can I
change Asterisk settings for soxmix (parameters)?
Regards, Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg
Sent: Tuesday, May 13, 2008 5:35 PM
To:
On Wed, May 14, 2008 at 10:41 AM, Stelios Koroneos
[EMAIL PROTECTED] wrote:
As people have sugested the ATX power supplies can work without a mobo
One thing to watch out for your setup is the actual ampere requirments for
your disks
i.e Your power supply provides 300W but this is
Hi
I've got a Dell with Intel Xeon and no Zaptel hardware installed.
However, as it needs to do IAX trunking and MeetMe conferences I need
timing enabled using ztDummy.
However, when enabling ztDummy, Playback() and Background() both fail to
play.
If I place NoOp(${PLAYBACKSTATUS}) after the
Hi,
exten = _0[23456789].,1,SetCallerPres(prohib)
did it for me.
Thank you,
Stefan
Andreas van dem Helge wrote:
On PRI SetCallingPres works fine it should work with ISDN because its
the same signaling.
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set CallerID
On Wed, May 14, 2008 at 10:15:01AM +0200, Koch Máté wrote:
Tim Panton wrote:
I think that if you use meetme, you will automatically drop to 8khz
sampling because that is what zaptel uses to do the mixing.
If you want wideband, you will probably need to make one-to-one calls.
That
Hello,
Andreas van dem Helge schrieb:
On PRI SetCallingPres works fine it should work with ISDN because its
the same signaling.
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set CallerID Presentation
[Description]
SetCallerPres(presentation): Set Caller*ID presentation on
Hello users,
This is regarding MeetMeAdmin() administration from DialPlan
exten = 12345,1,MeetMe(123|MX) ; Enter conference number 123
;Exit conference
by pressing a single digit
exten = 12345,2,Hangup()
exten = 1,1,MeetMeAdmin(123|M|1)
On Tue, May 13, 2008 at 7:31 PM, equis software [EMAIL PROTECTED] wrote:
What I need to configure in my * to permit make calls only registered sip
users??
Nothing. You can't call unregistered SIP users since you don't have
any contact information for them so therefore all your calls will only
On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff
[EMAIL PROTECTED] wrote:
I'll be doing a new Asterisk deployment soon, and would like to gather your
thoughts.
Here are some items that need to be kept in mind:
Support 800 phones (400 of which are analog)
Concurrent calls ... ? but need to
You can call the sox binary directly from your dialplan, or any other
binary that fits your needs. If you post your dialplan where you're
doing the recording, we can give input about where to put the calls to
sox.
On Wed, May 14, 2008 at 4:16 AM, Asterisk [EMAIL PROTECTED] wrote:
Thanks. If I
i think you have to have a mail transport agent like sendmail or postfix
installed and configured on your asterisk box , however if you forward the
mails to say hotmail or yahoo or gamil those servers will reject the mail
transfere
- Original Message -
From: Roberto Milani [EMAIL
It should already work, unles you configured your queue differently? :)
l.
On Tue, 13 May 2008 14:44:44 +0200, bilal ghayyad [EMAIL PROTECTED]
wrote:
Hi list;
Any one can advise how to put the caller in the queue
in case no one available to take his call? All are
busy (having calls)?
I would have to agree with Grey Man, a pilot project is one way to start up.
I would also seriously recommend buying some consulting time from an
experienced Asterisk PBX vendor/dealer/consultant.
The cost is negligible in light of the scope of your project.
A pilot project will only give you
I am about to order some DIDs for my first install but I am unclear on how
Asterisk
will function in either scenario with the two options I can order with. One
option
is the DID has unlimited connections. Another option for the DID is that it has
a
maximum of two concurrent calls only. How does
Ditto.
If you need to quantify the consultant to the powers that be just ask
for an Infrastructure Audit. I have done several in the past that
have saved tons of money that encouraged further phone projects.
Finding dead phone lines to discovering unused but rented telcom gear
is always fun.
Joseph
The DIDs are tied to a circuit. The circuit has a ring order
(ascending or descending or other...). So ordering the DIDs is just
getting the numbers most of the time, attaching them to a circuit that
is setup to handle the calls in a certain way.
Andrew
On Wed, May 14, 2008 at 9:57
Dear fellow Asterisk users,
Voiceroute is proud to announce the first North America Druid Meetups
happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27
May 2008)
Druid Meetups are basically fun demo sessions of Druid (Open Source
Edition Unified Communications Server). Come and
Ming,
Are you coming to New York? Would be great to have an Asterisk related
meetup here as well.
Regards,
Dean Collins
[EMAIL PROTECTED]
Cognation Limited
+1-212-203-4357
+61-2-9016-4652 (Sydney indial)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Date: Tue, 13 May 2008 22:28:33 -0400
From: OCG Technical Support [EMAIL PROTECTED]
Subject: Re: [asterisk-users] voicemail not sending emails
To: 'Asterisk Users List' asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Permissions? Try
I assume you are going to with a VOIP provider.
Essentially, you have one DID and any number of channels/ports.
Typically, you pay per port with a minute charge.
Some people give you unlimited ports but charge a higher per minute fee.
In you case, where you currently have 3 lines, you would
Ming Yong schrieb:
For our first meetups, we have the below cool stuff we will be demoing
third party integrations with Druid Asterisk using the Druid SOAP
API and how people can develop third party apps
shameless plug
Or if you happen to be located in Germany join us at Asterisk-Tag.org
On Wed, 14 May 2008, Roberto Milani wrote:
From: OCG Technical Support [EMAIL PROTECTED]
Permissions? Try running msmtp from the asterisk account? (Assuming
that is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
thanks for the replies but the
Steve Totaro wrote:
This looks like it may be your problem.
http://bugs.digium.com/view.php?id=9592
(0070069)
qwell - administrator
09-06-07 17:05
Closing.
The simple solution here is to just comment out the #define USE_RTC in
ztdummy.c. The ztxen module does not appear to be
The Asterisk.org development team has released Asterisk versions 1.4.20-rc3 and
1.6.0-beta9.
These releases are intended to encourage community testing to improve the
quality of the upcoming 1.4.20 and 1.6.0 releases. The testing process has
proven extremely useful and we would like to thank
Roberto - I noticed in your original email you had the lines
something like
mailcmd=/opt/local/bin/msmtp -t ; --from blah
AND
serveremail=from=blah
In mailcmd everything after the ; will be ignored as a comment
In serveremail - well - it should throw an error...
I would probably
I see.
So how does Asterisk assign Lines to the various channels?
I intend to have a few Aastra 480i's and these phones I believe
have 4 line buttons on them, does the functionality of Asterisk
in this scenario allow someone to see Line 1 is in use and either
pickup the phone and attach to a free
Tobias Wolf wrote:
Lets say i have a configured number range from 1000 to 1999 and 1000 is
my base number. I make an outgoing call from a phone which sets its
CallerID to 1500.
Can anyone be so kind to tell me what is shown to the callee in either
case?
I can only tell you, that after I
Do you have 30 to 50 people in New York?
We only tend to get about 10 people to the asterisk meetup events which
is disappointing.
Who else on the Asterisk list is based in NY that would like to catch up
31 May (6-9 pm) for a Druid event/General Asterisk event
Regards,
Dean
Those buttons are call appearances.
They function based on how the phone is configured and how you
program asterisk to process calls.
For example, you have a phone that is ext #101, you have 4 call
appearances on the phone device.
You receive 2 phone calls within the span of 2 seconds, 2 of
Anyone out there use Druid and can comment on it? I found out it was
once closed source by the fact that they announced that it is now
opensource.
Before that, I had never heard of it. Usually I pick up on chatter if
something is good.
I see there are only 19 threads in the forums and some are
- Original Message -
From: gres [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 14, 2008 4:23 PM
Subject: Re: [asterisk-users] voicemail not sending emails
i think you have to have a mail transport
Steve Totaro schrieb:
Anyone out there use Druid and can comment on it? I found out it was
once closed source by the fact that they announced that it is now
opensource.
Before that, I had never heard of it. Usually I pick up on chatter if
something is good.
I see there are only 19
Tried to install it on a dev box that had been running Trixbox. Kernel
panic midway through install. Happened twice in a row, so we gave up.
I've heard some people really like it, which is why we wanted to have a
look, but no joy for us.
N.
Steve Totaro wrote:
Anyone out there use Druid
On Wed, May 14, 2008 at 7:50 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Anyone out there use Druid and can comment on it? I found out it was
I don't use it per se, but afyter a conference with Voiceroute, I
promised to install it and I did so on a test box. The install was
great and the
On Wed, May 14, 2008 at 3:03 PM, randulo [EMAIL PROTECTED] wrote:
On Wed, May 14, 2008 at 7:50 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Anyone out there use Druid and can comment on it? I found out it was
I don't use it per se, but afyter a conference with Voiceroute, I
promised to install
You could do it that way but there is really no need. If you are
getting DIDs you can just have them ring a certain phone, a group of
phones, an application, a queue.. You can just abandon the whole
notion of Lines.
Thanks,
Steve Totaro
On Wed, May 14, 2008 at 12:44 PM, Joseph L. Casale
Hi,
I have read about SS7 recently and learnt that it is a signalling protocol
used in PSTN for call management, setup, etc. The thing that I don't
understand is how SS7 plays a role in VOIP. When I make calls between
landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it
I'm trying to convince my employer to deploy an Asterisk based system, but
one member of the leadership team is against it. The rest of the team is
for it, but he's convinced them that we should find other organisations in
the Joplin, MO area who are using Asterisk first because, we don't want to
Bryson Medlock wrote:
I'm trying to convince my employer to deploy an Asterisk based system, but
one member of the leadership team is against it. The rest of the team is
for it, but he's convinced them that we should find other organisations in
the Joplin, MO area who are using Asterisk first
Tell your Employer to have a little faith.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryson
Medlock
Sent: Wednesday, May 14, 2008 3:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] anyone from Joplin, MO
I'm
Roberto Milani wrote:
Roberto - I noticed in your original email you had the lines
something like
mailcmd=/opt/local/bin/msmtp -t ; --from blah
AND
serveremail=from=blah
In mailcmd everything after the ; will be ignored as a comment
In serveremail - well - it should throw
SS7 does NOT play a roll in VoIP. The SS7 signaling that you are
describing is not really SS7 but signaling over a PRI using ISDN that
your provider uses to exchange information via SS7 to the other
carriers.
To be blunt and I do not mean to be condescending in any way, but, if
you are using
does the /tmp directory need to have some specific kind of mode/
ownership?
mine is linked to /private/tmp and is lrwxr-xr-x root admin
Ciao
Roberto
On May 14, 2008, at 8:34 PM, Roberto Milani wrote:
That's what I thought,
and my voicemail.conf is:
[general]
format=wav
attach=yes
Aadil,
If the answers are not suitable for you, you might want to check out freeswitch.
Thanks,
Steve Totaro
On Wed, May 14, 2008 at 11:30 PM, Paul Hales [EMAIL PROTECTED] wrote:
Dear Aadil,
You asked this question about 1 month ago, and received several
response.
Were you unhappy with
First of all thanks to everybody
I feel the need to clarify the configuration.
from the command line msmtp works, this means that ~.msmtrc is
configured properly
I removed the mailcmd line from voicemail.conf , renamed sendmail to
sendmail.orig and created a link to msmtp called sendmail
Hello users,
i am trying to setup a conference system
and i have following requirement
1)some users are only in listen mode
2)some users are only in talk mode
3)some users are able to do both talk and listen
how to diffrentiate them when they enter into a particular mode?
meaning do i have to
gmail wrote:
Does anybody know how to off-load an Asterisk Box so that to distribute
its functions like IVR and VoiceMail or its PTSN gateway function into
different servers? in this case , will the installation of Asterisk on
each server differe and how these different servers will
Basic info site: http://VoipUsersConference.org
Hi,
What if you could connect people or businesses without having them
require hardware or software of any kind? I've always been interested
in this idea and now it's a reality with several choices to
investigate.
We've spoken to Yusuf Motiwala,
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