This arrived today: July 16th. (but I also receive older messages;
oldest is from july 4th)
Looking at the headers of the messages it always appears to be the last step
that is delayed. (that is the delivery to my local mail server).
Example below:
Received: from viadoos.rzuiderven.nl ([unix
Hi all,
My scenario is such that I have three users connected to a conference.
CLI meetme list 1234
User #: 01 9176502096 no nameChannel: Zap/23-1
(unmonitored)00:00:32
User #: 02 john john Channel: SIP/john-b7800468
(unmonitored) 00:00:28
User #: 03
1. Instead of using *1 (automon) I need to record each phone call at a
certain * box.
2. While already talking about this. I want to autodelete with cron at 2
am in the morning all recordings which are older than 50 hours! How can
I do that?
bye
Ronald
Hello,
I tried to configure a very simple case of Asterisk using SIP
userA --- Asterisk server userB
sip.conf
[userA]
type=friend
username=userA
host=dynamic
nat=no
context=test
[userB]
type=friend
username=userB
host=dynamic
nat=no
context=test
In extensions.conf
[test]
exten =
1) You can use mixmonitor to record all calls.
2) You can write a cron that will delete all calls that are older than X
amount of time.
How do you do it ? Simple - Google is your friend.
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Sun, Jul 15, 2007 at 10:36:19AM -0400, Robert Moskowitz said:
Joe acquisto wrote:
APC makes a two line unit. PTEL2. But it's two lines in one jack.
I have always been a fan of Triplite. They use old tech when
appropriate. I am big on Line Conditioners and UPSs with line
I do not need g723.1 codec, this is not the problem, here is another
description of the problem:
The client offer 2 codecs (g729 and g723) for all calls, my server accept
only g729, so normally the client server will negotiate the codec and both
sides agrees on g729, but this does not happened
Or you could just show him... :)
Heres one I added, I have a global variable defined (RECORDSIP), so that
I can switch the record on/off without having to hack the code all over
the place... This uses Monitor instead of mixmonitor as I only want one
file. In the dialplan I have:
exten =
Hi All,
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could have different users sitting there. Is
On Tue, 10 Jul 2007 13:15:20 -0500, The Asterisk Development Team wrote
The Asterisk development team has released Asterisk version 1.2.21.1 and
1.4.7.1. These releases are minor updates to the releases that were
made yesterday to fix a couple of introduced issues. One issue was
related to
On Sat, 2007-07-14 at 13:20 -0700, bilal ghayyad wrote:
[incoming]
include = parkedcalls
exten=103,1,Dial(SIP/Bob,,tT)
exten=104,1,Dial(SIP/Charlie,,tT)
When we use tT and when we use t alone or T alone, I
know this for call parking, but I do not know what the
tT does?
The 'tT' means use
On Sat, 2007-07-14 at 23:31 -0400, Forrest Beck wrote:
Has anyone come up with to timestamp a Recording? I am using a pretty
simple dialplan to record a audio file for a hotline. I'd like to
store the date and time it was recorded somewhere, Ast DB or MySQL DB.
In that case, the ${EPOCH}
On Sun, 2007-07-15 at 10:50 -0400, Tom Moore wrote:
What are the ideal settings I should use in iax.conf for the optimal
operation over satellite besides the normal options for the type=friend
peer?
If you have a suitable timing source (ztdummy or a Digium card) and more
than 2 simultaneous
Has anybody created a crontab script to check the health of an Asterisk
server?
The part I'm struggling with is some sort of IAX ping to test the
connection to each provider without making a call.
-HJC
___
--Bandwidth and Colocation Provided by
On Mon, 2007-07-16 at 00:07 -0700, Arpit Mehta wrote:
Also what does auto fall through mean ?
Auto-fallthrough happens when Asterisk runs out of priority numbers for
the current extension. For example, if my extension looks like:
exten = 123,1,Playback(hello-world)
exten =
On Mon, 2007-07-16 at 03:01 -0700, Tuan Viet Nguyen wrote:
I make a call from userA to userB, it works, but I have 2 questions:
1/ By verifing with Wireshark, I see that the CallID of the INVITE
message
sent from userA to Asterisk is different from the CallID of the INVITE
message sent from
Does anyone know if X-Ten or SJPhone support multiple cordless handsets for
multiple lines? I have an office with multiple roaming users(nurses) that are
in and out. I'd like to provide them telephones, and my idea is to have a PC
sitting in a corner somewhere running a softphone client.
On Mon, 16 Jul 2007, Henry J. Cobb wrote:
Has anybody created a crontab script to check the health of an Asterisk
server?
The part I'm struggling with is some sort of IAX ping to test the
connection to each provider without making a call.
According to your 'show hints' you've got hints setup OK but no watchers
- so it looks like it's your phones that are not playing as you would
like. Use a network analyser like WireShark to see if the phones send
out SIP SUBSCRIBE messages when you expect them to (e.g. at power-on).
Hi all, I use the G option in my dials for redirect both parties in the
conference.
There is a way for auto-include in a conference other parties that first
two without using AGI?
I try with:
[from-internal]
exten = ,1,Dial(IAX2/DIP02/||G(fromiax^^1)
[fromiax]
exten =
I have been looking for this solution for ages now... We have 5 hotels
using Micros-Fidelio POS and I really like to integrate Asterisk PBX.
Anything you can help...
Cheers,
Nitesh
Lee Jenkins wrote:
Tomislav Parcina wrote:
There is hotel application weary popular in Croatia -
Hello all,
I have seen some people asking how to configure asterisk to work with
h323 but i did not manage to do fix it yet (i am not an asterisk
expert).
Can someone help me configuring asterisk?
It is already compiled asterisk 1.4.5 with H323 support.
Everything looks fine.
Then i understand i
On Mon, 2007-07-16 at 09:31 -0500, Henry J. Cobb wrote:
Has anybody created a crontab script to check the health of an Asterisk
server?
The part I'm struggling with is some sort of IAX ping to test the
connection to each provider without making a call.
The VoIP wiki has a page[1] dedicated
Hello,
I have installed the latest beta of AsteriskNow on my machine.
Everything works fine except for one thing - my registration with
terminating peer times-out after some period of time.
I called my provider and they told that changing registration interval
should help.
I have once before
We have an issue with incoming calls from a provider in which DTMF tones
are sometimes sent using 'inband' and sometimes using 'rfc2833'. All
calls are G711 and the incoming SDP never indicates support for rfc2833.
Is there a setting in sip.conf that allows asterisk to receive DTMF
tones in
I will be out of the office starting 07/16/2007 and will not return until
07/26/2007.
For anything urgent pls. contact Fritz Reeve.
I will have limited email access but will have my cell phone 781-526-5513
if needed.
CONFIDENTIALITY NOTICE: This e-mail message, including any attachments,
is
On 7/16/07, Ronald Wiplinger [EMAIL PROTECTED] wrote:
1. Instead of using *1 (automon) I need to record each phone call at a
certain * box.
exten =
_1NXXNXX,1,MixMonitor(/var/spool/asterisk/monitor/${CALLERIDNUM}-${EPOCH}-${EXTEN}.wav)
exten = _1NXXNXX,2,Dial(Zap/R1/${EXTEN},90
Jeremy Mann wrote:
Does anyone know if X-Ten or SJPhone support multiple cordless
handsets for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I’d like to provide them
telephones, and my idea is to have a PC sitting in a corner somewhere
running
Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based
proxy / call routing setup? I need to get simple CDRs; not for detailed
settlement/rating, but just for reconciliation with an ultimate TDM
carrier just to make sure we only get billed for what we're actually
using.
I'd
Off topic, but involves an Asterisk deployment in a roundabout way.
Anyone here intimately familiar with Cisco Callmanager (Version 4-5),
that can tell me where a directory of the standard system voice prompts
for Callmanager might be obtained? I am looking for the text and
filenames of the
On 7/16/07, Cory Andrews [EMAIL PROTECTED] wrote:
Off topic, but involves an Asterisk deployment in a roundabout way. Anyone
here intimately familiar with Cisco Callmanager (Version 4-5), that can tell
me where a directory of the standard system voice prompts for Callmanager
might be
On Mon, 16 Jul 2007, Jeremy Mann wrote:
Does anyone know if X-Ten or SJPhone support multiple cordless handsets
for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I'd like to provide them telephones,
and my idea is to have a PC sitting in a
Hi,
I have a connection between a te120p and a siemens hipath which was recently
set up. The connection uses etsi / e1 with no crc. Every few hours the
asterisk shows the pri as in alarm
asterisk01*CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, In Alarm, Down, Active
But only when dialing bob or charlie. Only the second line, the
'include' line, is for call parking. The others are NOT for call
parking and are unrelated -- They are just for dialing charlie and bob
directly.
Jared Smith wrote:
On Sat, 2007-07-14 at 13:20 -0700, bilal ghayyad wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.18 and 1.4.4. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp
There are different h323 channel drivers. You seem to have built both. If
you try to dial using Dial(h323/.) then you need h323.conf. If you try
dialing using Dial(ooh323/.) then you need ooh323.conf. For me
personally the ooh323 (can't remember the name of it - sleep deprivation)
Hi Jeremy,
Sorry, I have no clue about your question, but I have a question in
regards to your USB Cordless handsets.
Do you have any idea on what you will be using ? We've tested a couple
of thems so far, but I'm still searching better products.
So if you have any ideas about any Wireless
Define health. I was working on but gave up on it (no time) to have serverA
call serverB. ServerB has an agi that it runs that stores info in DB. if
serverB doesn't get a call then we know that there are issues (and run the
script vice versa).
- Original Message -
From: Henry J. Cobb
On my CallManagers here at work, this appears to be what you're looking
for:
C:\Program Files\Cisco\TFTPPath\CCC\2_0_1_0\en_US
This is on CCM 4.1
Tim Reimers
Asheville City Schools
www.asheville.k12.nc.us
desk- 828-350-6195 mobile-828-545-3104
fax- 828-255-5454
Most days, there are
Just realised that that's the files... and you were asking for the text
read out in each one.
I don't know if there is any text of them available anywhere--
but the files are there and their names are there.
Tim Reimers
Asheville City Schools
www.asheville.k12.nc.us
desk- 828-350-6195
Thanks Tim, I managed to get someone to send me the actual prompts, and
I'm just going to have someone transcribe them and annotate the
filenames. I am also looking for the native prompts from Unity as well.
Appreciate the heads up!
Cory J Andrews
Executive Vice President
PBX Select, LLC.
On Mon, Jul 16, 2007 at 01:51:52PM -0500, The Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.18 and 1.4.4. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases
On 7/16/07, Dovid B [EMAIL PROTECTED] wrote:
Define health. I was working on but gave up on it (no time) to have serverA
call serverB. ServerB has an agi that it runs that stores info in DB. if
serverB doesn't get a call then we know that there are issues (and run the
script vice versa).
Do your SNOM phones sometimes use answer-after:0, and do
they have keyboard LEDs subscribed to their own extensions?
Do those people hangup calls by puttig down the handset
instead of pressing the X key?
We are seeing hanging channels in this particular case.
Ron
Michael J. Liberatore wrote:
Along the same lines -- Anyone know where I can get/extract the default
music on hold file from?
On 7/16/07, Cory Andrews [EMAIL PROTECTED] wrote:
Off topic, but involves an Asterisk deployment in a roundabout way.
Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that
Hello All,
I just updated my Asterisk server to 1.4.7.1, after reboot, I no longer hear
the audio for call to normal phones. Internal extension calls are ok. Has
anyone encountered the same problem? any thoughts?
Thanks
Nate___
--Bandwidth and
This may be more a Linux issue but maybe someone can help. I need to
know how to compile zaptel for a kernel version different from the one I
am running at the moment. Basically I need to do this because I run
into trouble when the customer updates the Linux kernel and the next
time they
Gordon Henderson wrote:
On Mon, 16 Jul 2007, Jeremy Mann wrote:
Does anyone know if X-Ten or SJPhone support multiple cordless handsets
for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I'd like to provide them telephones,
and my idea is
What type of Zap card?
Is this only on outgoing or only incoming calls or both?
On 7/12/07, Michael J. Liberatore [EMAIL PROTECTED] wrote:
Hi all, i am having a major asterisk problem. I think it started around
1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we
start
Hi All,
I have a site using Polycom 501 phones and for some reason the caller
ID of the phone number is coming up as sip:number@ip of server
Does anyone know why? It seems to be a Polycom thing as a Linksys phone
displays the CID number as just the number.
Hi David,
Disable URL dialing (url-dialing in the feature/ section of sip.cfg.
CP
Klaverstyn, David C wrote:
Hi All,
I have a site using Polycom 501 phones and for some reason the caller
ID of the phone number is coming up as sip:number@ip of server
Does anyone know why? It seems
On Mon, Jul 16, 2007 at 05:38:12PM -0500, Carlos Chavez wrote:
This may be more a Linux issue but maybe someone can help. I need to
know how to compile zaptel for a kernel version different from the one I
am running at the moment. Basically I need to do this because I run
into trouble
On Fri, 2007-07-13 at 14:12 +, lemmel lemmel wrote:
They used the 'local' dial option, with the D option (from memory)
--
Documentation :
D(digits): After the called party answers, send digits as a DTMF stream,
then connect the call to the originating
We have found that working WITH other Asterisk vendors is much more
pleasant than working against them - especially when you all run into
each other at a trade show.smile
PaulH
On Sat, 2007-07-07 at 11:04 -0400, Matthew Rubenstein wrote:
On Sat, 2007-07-07 at 08:39 -0500,
[EMAIL
Many thanks, working a beaute.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Tuesday, 17 July 2007 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CID on Polycom Phones
Hi David,
Disable URL dialing
Hello everyone,
I'm having problems with DTMF regeneration on my incoming PRI. I see that
when the calling party transmits a DTMF digit, it gets generated at my end
at a slower rate. Normally this would not be a problem, but several of
these calling party are alarm control panels that send
All you guys whining about delays for that past month: Ever thought the
problem is YOUR mail server? I have no problems at all. Messages arrive on
time.
On 7/16/07, Bill Maidment [EMAIL PROTECTED] wrote:
On Tue, 10 Jul 2007 13:15:20 -0500, The Asterisk Development Team wrote
The Asterisk
Until they rip off your IP, or just use all the public contributions in
combination with their better funded proprietary operation, without
contributing anything themselves, or even admitting they're using the
tech that could use the corporate boost.
FWIW, 3Com is not an Asterisk
Look at app_valetparking here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+addons
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Kiely
Sent: Monday, July 16, 2007 16:47
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/
Indicates support for Asterisk 1.4. The documentation listed suggests an
install like so:
cd /usr/src/asterisk
cp contrib/scripts/astxs /usr/bin/
cd apps
wget http://www.bkw.org/app_valetparking.c
cd ..
astxs -install
Andrew,
1) Please reply a topic on hand. If it is about the slow list then reply to
that thread. It is confusing to people if you reply to a release from the
Asterisk development team.
2) If you hav been following the list and the issue you will notice that the
problem is for a lot of users
Hello guys,
Does anyone has an Asterisk server hosted off-site ? Like in those data centers
that do web hosting in dedicated servers ?
Is there a hosting company that has a special plan to host voip services like
this, or usually is hosted in those dedicated servers like the ones I asked
On Mon, Jul 16, 2007 at 11:50:05PM -0400, Kevin Kiely wrote:
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/
Indicates support for Asterisk 1.4. The documentation listed suggests an
install like so:
cd /usr/src/asterisk
cp contrib/scripts/astxs /usr/bin/
cd apps
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