I was able to disable the DND button (no 9) on IP60x by putting the
following line in sip.cfg.
keys key.scrolling.timeout=1
key.IP_600.9.function.prim=Null/
However, I could not do so for the Services Button (no 29) on IP600 (or
Applications button on IP01)
keys
Hello Alejandro,
I'm not sure if it related, but I saw this behavior when the Asterisk
service was started without using the init script.
Regards
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, May 12, 2008
On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev
[EMAIL PROTECTED] wrote:
Hello All,
Is there a way to have Manager Bridge Channel to the specified extension
without the channel being connected.
In the current scenario the channel only bridges once the call get
connected, it does not bridge
Hello,
i´ve a problem i dont find the reason for. An incoming call coming over
iax is connected to a Sip phone. Until the phone picks up the call i
could hear moh without problems. Then the phone sets the call on hold
and opens another call to another extension. The incoming call hears the
2008/5/13 Stefan Schmidt [EMAIL PROTECTED]:
Hello,
i´ve a problem i dont find the reason for. An incoming call coming over
iax is connected to a Sip phone. Until the phone picks up the call i
could hear moh without problems. Then the phone sets the call on hold
and opens another call to
On Mon, 12 May 2008, Steve Totaro wrote:
You can put a TE405 in a 1 server (horizontally of course).
I've just built a bit of an experimental system with 2 PCI cards in a 1U
box. Quite a neat little system. The cards are a TDM400 and a TE120P. Will
they work? Well, they seem to, so-far... Be
2008/5/12 Steve Davies [EMAIL PROTECTED]:
Hi,
I read the WiKi, which implied there was a way of working around this,
but the HTML nature of the WiKi seems to have destroyed some of the
output so I cannot see the correct answer...
I would like to match a special case of a number dialled
If you want receive calls with the user B, must change the type to USER.
Please make tests and let we know the results.
Regards
On Tue, May 13, 2008 at 7:29 AM, gincantalupo [EMAIL PROTECTED]
wrote:
Hi,
I'm making some tests with Tellfree brazilian provider. I'm using 2
users A and B, one
Hi,
I'm having a little difficulty with my extensions setup.
What I'm trying to do is to have a PBX where I can call in to check
mail and call-out using the attached mobile or SIP phones.
If someone I know calls then they can be forwarded to me.
if it is someone I don't know then just ring the
On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Mon, 12 May 2008, Steve Totaro wrote:
You can put a TE405 in a 1 server (horizontally of course).
I've just built a bit of an experimental system with 2 PCI cards in a 1U
box. Quite a neat little system. The
Hello ,
is it possible to control multiple legs (channels) of a call
individually, ie.
call 1 -- incoming call connected to IVR
call 2 -- outgoing call to party a made via manager interface
call 3 -- outgoing call to party b made by call-script
I would like to allow the caller on call1 to be
Hi Steve;
If we give a look for the link
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
there are some topics not cleared to complete the
paging senario:
1) Why to use Set(_ALERT_INFO=RA)? And how can I
know what each device take?
2) Does the following work for Polycom:
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)?
Regards
Bilal
___
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asterisk-users
On Tue, May 13, 2008 at 8:35 AM, David Boyd [EMAIL PROTECTED] wrote:
Hello ,
is it possible to control multiple legs (channels) of a call
individually, ie.
call 1 -- incoming call connected to IVR
call 2 -- outgoing call to party a made via manager interface
call 3 -- outgoing call to
On Tue, May 13, 2008 at 4:22 AM, Grey Man [EMAIL PROTECTED] wrote:
On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev
[EMAIL PROTECTED] wrote:
Hello All,
Is there a way to have Manager Bridge Channel to the specified extension
without the channel being connected.
In the current
Hello users,
Actually i am planning to setup a conference system
i have following dialplan
[default]
exten = 12345,1,MeetMe(1234|X)
exten = 12345,2,Hangup()
exten = 1,1,MeetMeAdmin(1234|M|user1)
exten = 1,2,GoTo(12345|1)
exten = 2,1,MeetMeAdmin(1234|m|user1)
exten = 2,2,GoTo(12345|1)
exten
Hi Gabriel,
it works! I have tried FRIEND in previous tests but probably it did not
work because of other mistakes which disappeared when I changed the
parameters...
Thank you!!!
Giorgio
Gabriel Lopes wrote:
If you want receive calls with the user B, must change the type to USER.
Please
On Tue, 13 May 2008, Steve Totaro wrote:
On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Mon, 12 May 2008, Steve Totaro wrote:
You can put a TE405 in a 1 server (horizontally of course).
I've just built a bit of an experimental system with 2 PCI cards in a
On Tuesday 13 May 2008, Steve Totaro wrote:
You can be shot several times and not die. I would try
resetinterval=never just to be able to to say Not the problem
rather than Probably not the problem.
I'll do that, although I'm pretty sure that the setting is not the
problem as the yellow alarm
Florian Hackenberger wrote:
On Tuesday 13 May 2008, Steve Totaro wrote:
You can be shot several times and not die. I would try
resetinterval=never just to be able to to say Not the problem
rather than Probably not the problem.
I'll do that, although I'm pretty sure that the setting is not
Hi guys,
I have few queues configured on my PBX that have:
monitor-format=wav
monitor-join=yes
My problem is that the volume of the clients who are calling into these queues
is slightly higher than the volume of the agents.
Is there any way to modify the volume (either lower the volume of the
Steve Davies schrieb:
I found the same issue, and a similar issue with transferring a call
received out of a queue. Both issues exist in 1.2, and a friend of
mine kindly re-checked this and found that it was fixed/changed in 1.4
Regards,
Steve
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the
clients, or increase the volume of the agents) while doing the join of the
-in and -out files into one recording?
Uh-huh. Read the documentation for
On Tue, May 13, 2008 at 9:57 AM, Florian Hackenberger
[EMAIL PROTECTED] wrote:
On Tuesday 13 May 2008, Steve Totaro wrote:
You can be shot several times and not die. I would try
resetinterval=never just to be able to to say Not the problem
rather than Probably not the problem.
I'll do
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
the LAN behind the firewall.
Asterisk-Tag.org 2008
May 26th/27th
Berlin, Germany
http://www.asterisk-tag.org
http://www.heise.de/open/Marc-Spencer-eroeffnet-den-Asterisk-Tag--/news/meldung/107462
Speakers:
* Mark Spencer (Founder of Digium and Inventor of Asterisk) - Digium
* Kevin P. Fleming (Director of Software
-- Forwarded message --
From: Tilghman Lesher [EMAIL PROTECTED]
Date: Tue, May 13, 2008 at 11:20 AM
Subject: Re: [asterisk-dev] Paging intercom extensions
To: Asterisk Developers Mailing List [EMAIL PROTECTED]
On Tuesday 13 May 2008 10:05:19 Gideon Spreeth wrote:
The problem
Roderick A. Anderson wrote:
Can I, and if so how do I, set-up sip.conf to force my soft-phone to go
to a specific context when I take it off-hook?
This can be done with a analog phone, but I don't believe you can do it
on a sip channel.
(The [Dial/Answer] button
in ZoIPer). Or should I
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 guess high so that the server can handle
this.
Voicemail will be required
Problem solved turning off echo cancellation.
Any known bug?
Pepe Aracil escribió:
Hello.
I'm using asterisk in alarm reception system.
The system is DTMF intensive and works well while
all concurrent channels are online. But when one
channel goes hangup the other channels lose tones
I am a bit desperate trying to solve this problem. Sorry if I am
abusing the list a bit with the same king of question.
The problem I am having is very specific which is why it is very
difficult to diagnose and fix. Basically an Asterisk server is
connected via E1 PRI to an
Roderick A. Anderson wrote:
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
I have never seen a SIP aware firewall work with localnet and
externip/externhost. You should try either disabling the SIP fixup on
your firewall or remove the localnet/externip from sip.conf.
Carlos Chavez wrote:
I am a bit desperate trying to solve this problem. Sorry if I am
Gordon Henderson wrote:
On Tue, 13 May 2008, Steve Totaro wrote:
On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Mon, 12 May 2008, Steve Totaro wrote:
You can put a TE405 in a 1 server (horizontally of course).
I've just built a
I have a queue with the following setting.
total queue member =30, autofill=1, timeout=25, monitor_format=wav49
asterisk 1.4.18
In busy hour, the loading of CPU reaches over 300%. At that moment,
all members are occupied and many calls are waiting in the queue.
There will be choppy and line cut
Andreas van dem Helge wrote:
A quality 3U chassis will mount the cards parallel to the mainboard
with the use of a riser card, just as a 1U chassis does.
If you are intent on sourcing the components yourself may I suggest a
Tyan or Supermicro barebones server? I think that is the best
Have you tried GetVariableCommand and GetFullVariableCommand?
See
http://asterisk-java.org/development/apidocs/org/asteriskjava/fastagi/co
mmand/GetFullVariableCommand.html.
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sherwood McGowan
On Tue, 13 May 2008, Steve Totaro wrote:
Gordon,
Did you hijack someone else's thread? You should really start your own.
Not really - I was replying to the 3U/1U thread and rambled on a bit ...
Anyways, I have had great luck with TDM card to TDM card faxes on the
same box.
I think you
The Asterisk.org development team has released Asterisk version 1.4.19.2.
This release includes some IAX2 channel driver updates. Asterisk 1.4.19.1 was
released to address an IAX2 security vulnerability. Unfortunately, the changes
to address the security issue had an unfortunate negative
What I need to configure in my * to permit make calls only registered sip
users??
Thanks!
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To UNSUBSCRIBE or update options visit:
Do you mean
What do I need to configure on my * installation so that only registered sip
users can make calls? ?
If so, you are going to need to give a lot more details regarding your current
configuration for you to get any answers.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL
On Tue, 13 May 2008, equis software wrote:
What I need to configure in my * to permit make calls only registered sip
users??
1) RTFM
2) Google
3) http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
4) Ask again with more detail in the body and more specificity in the
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.
One of the features that will be important (particularly for the
receptionist desk is
Steve Totaro wrote:
Roderick A. Anderson wrote:
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a
[EMAIL PROTECTED] schrieb:
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.
One of the features that will be important
We use the linksys 942's and they work flawlessly and are easy to setup
The CISCO phones do not come with SIP, you have to upgrade their
firmware from a TFTP server.
[EMAIL PROTECTED] wrote:
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is
that they support a max of 50 BLF subscriptions... you can setup up to 180 blf
keys with 3 560Ms but it will still only subscribe to a max of 50... from what
I understand it's a firmware limitation.
For 4-6 phones
One of the features that will be important (particularly for the
receptionist desk is to show status of the other lines in use). I
don't
want the receptionist to pick up a line if it being used.
Hi Dayton,
It's even easier than that. With an asterisk PBX your receptionist
shouldn't be
Hi group
I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla
Schroder to make my first call. My asterisk box is on a Debian box with an
public static IP. The clients (2) are with dynamic private IP's
I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between
I am about to start my first Asterisk installation and will only
be using IP phones (either Snom's or Aastra's), I have a local voip
Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this
up inside Xen on a CentOS pv guest. I understand from reading old posts
since I am not
Bill Michaelson wrote:
Alex Balashov wrote:
Steve Totaro wrote:
This make more sense:
Open WiFi AP (or cracked WEP) hacked Asterisk box (who sets the
CID/ANI Telco -- terminated to the PSTN
Well, sure, but you can do far worse things than spoof ANI/CID with
On Tue, May 13, 2008 at 4:56 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
I am about to start my first Asterisk installation and will only
be using IP phones (either Snom's or Aastra's), I have a local voip
Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this
up inside
Joseph L. Casale wrote:
I am about to start my first Asterisk installation and will only
be using IP phones (either Snom's or Aastra's), I have a local voip
Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this
up inside Xen on a CentOS pv guest. I understand from reading
Sure if you don't need ztdummy, or is there a newfangled way around that?
Thanks,
Steve Totaro
Hi Steve,
I read the wiki and see this provides timing for Asterisk. Can you point
me toward a description of what exactly this does? I was checking out the
tutorial at
On Wed, May 14, 2008 at 01:27:38AM +0800, Rilawich Ango wrote:
I have a queue with the following setting.
total queue member =30, autofill=1, timeout=25, monitor_format=wav49
asterisk 1.4.18
In busy hour, the loading of CPU reaches over 300%. At that moment,
all members are occupied and many
On Tue, 13 May 2008, Alex Balashov wrote:
Joseph L. Casale wrote:
I am about to start my first Asterisk installation and will only
be using IP phones (either Snom's or Aastra's), I have a local voip
Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this
up inside Xen on a
SPA942s do not currently support BLF keys. The four lit buttons are
line keys only with the current firmware, although our Linksys rep has
assured us that it's a feature to be supported soon.
John Signorello wrote:
We use the linksys 942's and they work flawlessly and are easy to setup
The
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
I was just following the wiki and installing Zaptel under my CentOS 5.1 Xen DomU
and after starting the service, the vm crashed. Now when restarting it, I get
the following.
Any ideas?
Thanks!
jlc
Kernel BUG at kernel/timer.c:331
invalid opcode: [1] SMP
last sysfs file:
On Tue, May 13, 2008 at 3:50 PM, [EMAIL PROTECTED] wrote:
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.
One of the features
On Tue, May 13, 2008 at 7:51 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
I was just following the wiki and installing Zaptel under my CentOS 5.1 Xen
DomU
and after starting the service, the vm crashed. Now when restarting it, I
get the following.
Any ideas?
Thanks!
jlc
Kernel
On Tue, 2008-05-13 at 16:24 -0600, Joseph L. Casale wrote:
Sure if you don't need ztdummy, or is there a newfangled way around that?
Thanks,
Steve Totaro
Hi Steve,
I read the wiki and see this provides timing for Asterisk. Can you point
me toward a description of what exactly this does?
Regulation, laws, and controls are NOT the answer. I like the freedom I
am entitled to, even with the Patriot Act. It will be a sad, sad day
when all thoughts, conversations, and transactions are logged and once
logged can be a form of control rather than a form of safety.
I have over a half-dozen different SATA hard drives, each with
different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each
one's different user groups and applications. Each one's load on the
Asterisk server is small enough that one server can host them all,
accessed easily over
On Tue, May 13, 2008 at 9:57 PM, Alexander Lopez [EMAIL PROTECTED] wrote:
Regulation, laws, and controls are NOT the answer. I like the freedom I
am entitled to, even with the Patriot Act. It will be a sad, sad day
when all thoughts, conversations, and transactions are logged and once
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote:
I have over a half-dozen different SATA hard drives, each with
different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each
one's different user groups and applications. Each one's load on the
Are you certain Asterisk is not sending the emails, rather than them not
being received? i have had problems in the past with spam filters
rejecting the emails.
On Tue, May 13, 2008 at 4:48 PM, Roberto Milani
[EMAIL PROTECTED] wrote:
Hello list users
I have a very nice installation of
I thought that the point that you had to have a timing source for *.
That source could be the clock off the T-1.
But if you didn't have something like your T1 to provide master clocking
ztdummy was something to provide the required a source for timing.
Joseph L. Casale wrote:
Sure if you don't
Helpful?
http://lists.digium.com/pipermail/asterisk-users/2005-April/097548.html
Thanks,
Steve Totaro
On Tue, May 13, 2008 at 10:28 PM, OCG Technical Support [EMAIL PROTECTED]
wrote:
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I
You don't need it except for a few applications such as meetme and IAX2..
I have come to always put some sort of timing hardware in a system
because ztdummy can be flaky under high use. A TDM400P with and FXS
module is usually what I suggest for fax, emergency phone, or whatever
else. It works
On Tue, 2008-05-13 at 22:46 -0400, Steve Totaro wrote:
On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED]
wrote:
I have over a half-dozen different SATA hard drives, each with
different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each
one's
At 11:45 PM on 13 May 2008, Matthew Rubenstein wrote:
The drives are 750GB drives, each one a different related set
of apps from a different Asterisk machine. I've consolidated them all
into a single Asterisk server. And I already have the existing PC
chassis and power supply, as well
CAUTION: doing this could be bad, i take no responsibility etc etc
Put a paper clip (or any join) between the green wire and any of the
black wires on an ATX power supply main lead to power it up without a
motherboard - google power up atx supply without motherboard if you
don't trust me
To turn on an ATX power supply that isn't connected to a motherboard use
a wire or paper clip to short the green wire (PS_ON) to any one of the
black wires (COM).
Pins 14 and 15
Now that's the cheapest solution I can give you
Alex
Snip...
If I
If I understand right, your problem is that the power supply won't turn on ?
ATX power supplies can be told to turn on by jumpering 2 pins on the
motherboard power connector. From memory its the Green wire and one of the
black wires, I usually use the next one inwards. Pinouts for the connector
On Wed, 2008-05-14 at 14:06 +1000, Col Ferguson wrote:
If I understand right, your problem is that the power supply won't turn on ?
ATX power supplies can be told to turn on by jumpering 2 pins on the
motherboard power connector. From memory its the Green wire and one of the
black wires, I
This will work:
http://www.newegg.com/Product/Product.aspx?Item=N82E16899705001
I assume you have devised a way to power the USB to serial adapters
from the PC power supply.
FWIW I think your system is inefficient but maybe you do need 750gb
per each installation. Each to his own.
On Tue,
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