On 00:05, Mon 31 Mar 08, Al Baker wrote:
Could you elaborate a bit more on :
For example, if I install zaptel from source, your support contract
with them is void.
Does this mean it is impossible to run Asterisk on Vendor Supported
versions of RedHat or Suse ?
Installing zaptel from
Thank you for all your time om your most detailed response.
It is extremely helpful.
The vendor's web page is
http://www.penguincomputing.com/index.php?option=com_contentid=170Itemid=209task=viewsysid=10007609
*PCI EXPANSION SLOTS*
Number of Slots 5
Slot Speed PCI Express: two x8
On Mon, Mar 31, 2008 at 01:04:53AM -0400, Al Baker wrote:
There are people who will support your Debian / Centos / whatever boxes.
If it is OK to ask on a non-commercial list, do you have a list of
reliable O/S support folks.
By this I mean companies with a support staff, as opposed to a
Hi Matt,
As you said, is this will work like this?
1. Student A will login in a conference room no 7789
2. Student B will login in a conference room no 7789
3. Student C will login in a conference room no 7789
4. Instructor for student A,B and C will login in a conference room no. 6689
5. When
Darrick Hartman (lists) wrote:
snip /
I didn't find it too much trouble in a Via C700N system. But I wouldn't
use one of the mainstream distros for the OS. They chew up system
resources just trying to accommodate any hardware.
The solution is to roll-your-own. See this series of articles
Hi,
I found out that GoTo in applicationmap is not working.
OK, LOCAL is working.
but I expected that applicationmap is using the DIAL option tT.
But it doesnt, it works without tT Option, so also callee can use internal
functions if callee knows the code.
Any workaround avaiable?
best
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I have a problem with Asterisk 1.4.x and the call manager. When I
originate a call by the call manager or by a dot-call file only the
calling party can hear the called party, not vice versa. When I dial the
same number directly from
Apart from the tutorial itself, what I wanted to point out was that the
way asterisk, zaptel and libpri are to be built is different for each
project, and this is sub-optimal; and that by building Asterisk as
required, you get a linkage error.
l.
On Sat, 29 Mar 2008 12:03:59 +0100, Alan
Hi,
Sometimes, you need to send requests to SIP phones either from Linux command
line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print $3}'
I'm wondering if
Good morning,
we face a problem with Atserisk 1.4.18.1 and Zaptel 1.4.9.2: calls are
frequently ended during conversation or voicemail are not registring the
entire messages given by callers.
What we have -and seem strange- is:
Module Size Used by
ztdummy
On Mon, Mar 31, 2008 at 09:11:00AM +0100, Alan Lord wrote:
Darrick Hartman (lists) wrote:
snip /
I didn't find it too much trouble in a Via C700N system. But I wouldn't
use one of the mainstream distros for the OS. They chew up system
resources just trying to accommodate any hardware.
2008/3/31, Simon Elliston Ball [EMAIL PROTECTED]:
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
That's fine !
Too bad one cannot input more specific database queries such as database
get
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to
The asterisk database system is really more of a hash table than a
full database, so it's unlikely to happen. It's actually berkeley db
underneath.
Of course you could always create your own table on calls by using
something like Set(DB(ips/692)=${SIPPEER(692|ip)}) in the dialplan,
but
Tzafrir Cohen wrote:
snip /
You can easily take a standard distro and remove all the services you
don't really need.
Yes, but you can't easily change the way the apps are built or setup,
e.g. compiler optimisations, use of initrd when not necessary, kernel
bloat just to accommodate any
Matthew Gibson wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
then in your sip.conf
[ext]
...
;secret=123
md5secret=MD5SECRET
Hey Martin, thanks for your response... Still no dice:
Quick questions... Where are the following coming from? Is this
something you
Hi All,
I am trying to establish a call between two users [EMAIL PROTECTED] and
[EMAIL PROTECTED] using ENUMLOOKUP. The following is my configuration.
In the DNS for domain1 I have the following entry.
5.4.3.2.1.domain1.com. IN NAPTR 100 10 u sip+E2U
!^(.*)$!sip:[EMAIL PROTECTED]
Hi,
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our end reliably.
If we originate the call, we can hang it up. This suggests to me that
the Sangoma A200D is
Hi there,
i'm beginner in Asterisk.. I have situation:
when a caller ,some SIP account on local network, dial outside number
trough some Zap channel, asterisk have to automaticly transfer call to
another local SIP channelIn other words, he have to change
caller..
For example:
I have SIP
If we accept a call originated elsewhere, then we cannot hang it up.
Only the call originator seems to be able to do that. The upshot is
that if asterisk hangs-up a line, and then tries to re-use it for an
outbound call before the remote has disconnected, we are simply
re-connected to the
Hi Ali,
On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote:
Hi All
I am developing a client that uses libjingle to do xmpp stuff with
ejabberd. I can also make audio calls between those clients. What I am
trying to archive now is to send calls to pstn using jingle. I was
SIP response 486 Busy Here is returned unless a divert contact is set
up in the phone config. I did a search through the SIP 3.0 Admin Guide
and didn't see any way of returning a different SIP response.
Dear friends,
I am trying to configure Asterisk so that it play differnt set of voicemail
greets for differnt extensions.
I put my customized .wav files under the extension, but it still does not
work. Asterisk still plays the default voice file.
On Mon, 2008-03-31 at 03:04 -0500,
[EMAIL PROTECTED] wrote:
Date: Mon, 31 Mar 2008 07:55:08 +0300
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Tests in VMWare
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our end reliably.
If we originate the call, we can
So should I register directly on the asterisk server or should I send
the voice calls through ejabberd to asterisk ?
On Mon, Mar 31, 2008 at 4:55 PM, Philippe Sultan
[EMAIL PROTECTED] wrote:
Hi Ali,
On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote:
Hi All
I am
Dear all,
I noticed a very strange problem. When I tried using VoiceMailMain to
record my unavailable message, the file does not get created even though I
can find the corresponding mssage from asterisk:
-- SIP/2001-b6307d78 Playing 'beep' (language 'en')
-- x=0, open writing:
All too common and largely undocumented. I had this same problem.
Installing ztdummy changes Asterisk to use it for timing of playback,
apparently. Removing ztdummy fixed the problem. To get it all to
work, I had to upgrade to to at least kernel 2.6.23.11 (previous
versions are either missing
Any new about this?
Thanks
On Thu, Mar 27, 2008 at 11:29 AM, equis software [EMAIL PROTECTED]
wrote:
I was trying to trap SIGHUP, but could be another signal because it didn't
work.
I'm doing this
class MyScript():
def logsignal(self,signum, frame):
On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote:
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our
On Mon, Mar 31, 2008 at 4:51 PM, Ali Jawad [EMAIL PROTECTED] wrote:
So should I register directly on the asterisk server or should I send
the voice calls through ejabberd to asterisk ?
You can't register an XMPP client on Asterisk, because it's not an
XMPP server. The required steps to
On Mon, 2008-03-31 at 16:25 +0100, Steve Davies wrote:
On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote:
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most
Does AMD (answering machine detect) need ztdummy or some other timer to
function properly?
--
Drew Miller
Iowa Democratic Party
Information Technology Director
Office: (515) 974-1682
Cell: (515) 451-4509
AIM: ItsDrewMiller
MSN: [EMAIL PROTECTED]
On 31/03/2008, David Boyd [EMAIL PROTECTED] wrote:
You should ask for ground start signaling. This will resolve your
issues.
Could you point me at some reference material for how this differs
from KS, and what compatibility issues this might cause with other
equipment? Has anyone tried this
On Mon, Mar 31, 2008 at 11:08:09AM -0400, Norman Franke wrote:
All too common and largely undocumented. I had this same problem.
Installing ztdummy changes Asterisk to use it for timing of playback,
apparently. Removing ztdummy fixed the problem. To get it all to
work, I had to upgrade to to
No It does not require.
Regards,
Sanjay.
- Original Message -
From: Drew Miller [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] Simple Question
Does AMD (answering machine detect) need
Hello,
I am trying to test-call my own asterisk server to see if I can
receive SIP calls properly.
I use a softphone to call the SIP address, and because twinkle
doesn't support SRV records, I go via a proxy.
When the call comes in, asterisk says:
handle_request_invite: Sending fake auth
On Mon, 31 Mar 2008, Alan Lord wrote:
Also, can you
find 300Gb of solid state storage for about £30. ;-)
Where??
Gordon___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
On Mon, 31 Mar 2008, Steve Davies wrote:
On 31/03/2008, David Boyd [EMAIL PROTECTED] wrote:
You should ask for ground start signaling. This will resolve your
issues.
Could you point me at some reference material for how this differs
from KS, and what compatibility issues this might cause
Announcement:
We are pleased to announce that we have released AsterPas FastAGI ObjectPascal
Script Server for Asterisk PBX under license.
What is AsterPas?
AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX
call flow using ObjectPascal based scripting.
Dear list, this is getting ridiculous - I would read the specs and
compare
From the link below it says:
* Two Available PCI Express x8 Slots
* Two Available PCI Express x8 Low Profile Slots
* One Available 64-bit/100MHz PCI-X slot
The list has already answered what goes in what
Lee Jenkins wrote:
Announcement:
We are pleased to announce that we have released AsterPas FastAGI ObjectPascal
Script Server for Asterisk PBX under license.
Oops. That should be LGPL license ;)
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote:
I have, on many occasions, had kernel
panics when trying to shut down wanrouter. I don't have this 'fear'
with Digium cards.
I never have had those issues if you don't execute zaptel init.d
script, because it tries to
Hello,
Is it possible to request for the premission from the called party through
a prompt before routing the call?
For instance, before actually connecting two parties through the use of DIAL
command in the dialplan, I want to let Asterisk to automatically
ask for the called party to decide
Octavio Ruiz wrote:
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote:
/etc/sysctl.conf:
kernel.panic = 1
OR
echo 1 /proc/sys/kernel/panic
OR
pass panic=1 as a kernel parameter in your grub.conf/lilo.conf
Now that is nice to know, thanks!
Doug
--
Ben
Dear all, I have an Asterisk 1.4.13 with SIP protocol and several voip
clients (Twinkle, X-Lite and SJPhone). Every call among voip clients
pass through the Asterisk server, so there isn't any voip packet
client-to-client.
Can Asterisk control the RTP open ports the voip clients use ??? Or the
We have just implemented cdr-custom. Works fine minus the timestamps that
appear in the CDR.
The system's timezone is GMT. In the configuration usegmtime=yes is set.
However, all of the CDRs in the Custom CDR comes as GMT-5.
Another oddity is that the standard cdr/Master.csv is using
On Mon, Mar 31, 2008 at 11:54:14AM -0600, Octavio Ruiz wrote:
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote:
I have, on many occasions, had kernel
panics when trying to shut down wanrouter. I don't have this 'fear'
with Digium cards.
I never have had those
Yes it is.
I'm remote at the moment so I can't send you the code but google for mobile
remote receiver and you'll find what you are looking for.
Lots of people do it so they don't have calls to cell phones picked up by
voicemail.
Cheers
dean
-Original Message-
From: Pete Kay [EMAIL
Gordon Henderson wrote:
On Mon, 31 Mar 2008, Alan Lord wrote:
Also, can you
find 300Gb of solid state storage for about £30. ;-)
Where??
Gordon
Sorry my bad. It was a question...
Al
--
The way out is open!
http://www.theopensourcerer.com
On Mon, Mar 31, 2008 at 12:58 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
There are a number of
ways to order that. In most distributions it is done by explicit
ordering. In some it is done by dependencies.
Gentoo is one of them, where if you run directly from CLI
/etc/init.d/zaptel stop and
I will be out of the office starting Mon 03/31/2008 and will not return
until Tue 04/01/2008.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Tzafrir Cohen wrote:
On Mon, Mar 31, 2008 at 11:54:14AM -0600, Octavio Ruiz wrote:
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote:
I have, on many occasions, had kernel
panics when trying to shut down wanrouter. I don't have this 'fear'
with Digium cards.
I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S).
Does anyone have a valid XMLDefault.cnf.xml they could share?
I have tried the version at
voip-infoinfo.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=14768#Troubleshootingfor
the
On Mon, 31 Mar 2008, Alan Lord wrote:
Gordon Henderson wrote:
On Mon, 31 Mar 2008, Alan Lord wrote:
Also, can you
find 300Gb of solid state storage for about £30. ;-)
Where??
Gordon
Sorry my bad. It was a question...
Ah, Doh... I mis-read it all.. Curse my dyslexia!
However I've
Razza wrote:
I have 7965 and am trying to convert the firmware to SIP
(SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they
could share?
I have tried the version at voip-info
I'm looking to install a system with 80 FXS analog phones.
At this time the only cost effective solution is using a 4 port T1 card and
addit 600 channel bank.
Has anyone tried this solution? any good documents beside
http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote:
YMMV Change to reflect your firmware (e.g. P003-07-4-xx)
8 SNIP 8
I removed the following lines:
loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8
loadInformation7 model=IP Phone
On Mon, 2008-03-31 at 23:07 +0100, Razza wrote:
On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote:
YMMV Change to reflect your firmware (e.g. P003-07-4-xx)
8 SNIP 8
I removed the following lines:
loadInformation8 model=IP Phone
7940P003-07-4-00/loadInformation8
Al lists wrote:
I'm looking to install a system with 80 FXS analog phones.
Each channel bank can handle 48 analog channels, 2 PRIs per box.
as far as i know, addit 600 T1 interface is not PRI (please correct me
if i'm wrong) its CAS robbed bit, will that work with new Digium quad
T1 like
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
Al lists wrote:
I'm looking to install a system with 80 FXS analog phones.
Each channel bank can handle 48 analog channels, 2 PRIs per box.
as far as i know, addit 600 T1 interface is not PRI (please
correct me
if i'm wrong) its CAS
Hello everybody, i'm from Mexico, at the time i´m working on a production
server with asterisk 1.2.25 + spandsp-0.0.4 +
libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I
installed this version of astunicall that i downloaded from
http://www.moythreads.com/astunicall/
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote:
I'm looking to install a system with 80 FXS analog phones.
At this time the only cost effective solution is using a 4 port T1 card and
addit 600 channel bank.
Has anyone tried this solution? any good documents beside
It can be done via the 'visit a macro' part of the dial command...
If anyone would like, i can post a code sample.
PaulH
On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
Yes it is.
I'm remote at the moment so I can't send you the code but google for mobile
remote receiver and you'll
Don Pobanz wrote:
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
This does not sound right. If it is 2 PRIs then it should be 46 channels
I may have the terminology incorrect. I don't have a D channel, so I
guess this would be called a T1 then?
Doug
--
Ben Franklin
Doug Lytle wrote:
Don Pobanz wrote:
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
This does not sound right. If it is 2 PRIs then it should be 46 channels
I may have the terminology incorrect. I don't have a D channel, so I
guess this would be
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before
Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
Steve, what are my options for SIP to fxs?
thank you!
On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote:
Don Pobanz wrote:
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
This does not sound right. If it is 2 PRIs then it
Something like this:
Dialling:
exten = s,n(dial),Dial($ZAP/G1/${number},15,M(check)gm)
exten = s,n,Dbget(next/number)
exten = s,n,Goto(dial)
{macro-check}
exten = s,n,Playback(${heresacall})
exten = s,n,Read(response,options,1)
exten = s,n,Goto(${response},1)
exten = 1,1,Macroexit
exten =
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