Asterisk-1.6.0.3
OS-2.6.24.2.dn.p4
kernel-CentOS release 4.6 (Final)
libpri-1.6 compatable
zaptel-1.6 compatible
I have been using the accounts for faxin for faxing.
For some of the numbers when i send fax it went through successfully.
For some numbers the following error is occuring in asterisk
Kevin P. Fleming kpflem...@digium.com wrote:
Trevor Hammonds wrote:
New development: I've assigned an external DID to the fax extension,
and fax calls come in fine, with a strange burst of noise about one
second into the preamble. However, I am still unable to transfer an
inbound call
I have a little problem that i can't solve by myself:
My LDAP server is OK, i'm sure of that.
All my LDAP users register without any problem.
My goal is to retrieve the name of the Queue for a person, but when
LDAPget is called, I have no output in the asterisk CLI and it hangs up...
Thnx Mark,
but there is no such option for 941 in 5.1.8. As far as I know,
5.1.8 is the lastest I can get for this phone.
I also tried to enable TLS with SIP_Transport_1_
ua=naTLS/SIP_Transport_1_
and Proxy_1_ ua=roxxx.xxx.xxx.xxx;transport=tls/Proxy_1_
in the config file(s), but it didn't work.
I attached the show channels in my first post, but removed it to reduce the
length of replies. Here it is again along with show status.
Note that there is only 1 PRI currently attached.
geriatrix*CLI dahdi show status
Description Alarms IRQbpviol
CRC4
On Tue, 19 May 2009, Olivier wrote:
Hi,
In ASTDB, I've got a rather long list of entries like:
/FamilyA/Key1Value1
/FamilyA/Key2Value2
/FamilyA/Key3Value3
...
Instead of sending several DBGet queries (and parsing every response), I'm
wondering if a single database show or
Dear List members.
I want suggestion to use digium TC400 or can I
use G.729a/b codec with out card?
If yes then any difference in both scenario.
Best regards,
Kashif Ali
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-- Bandwidth and Colocation Provided by
2009/5/20 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 19 May 2009, Olivier wrote:
Hi,
In ASTDB, I've got a rather long list of entries like:
/FamilyA/Key1Value1
/FamilyA/Key2Value2
/FamilyA/Key3Value3
...
Instead of sending
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Days ago I bought a OpenVox A400P card with a port FXS and another FXO
that I am testing with my Asterisk installation in my house. I'm using
Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on
Debian GNU/Linux Lenny.
I was
Steve Totaro stot...@first-notification.com writes:
I thought it was common knowledge that the queue app in Asterisk has this bug.
I have recordings to prove what I originally dismissed as impossible
or user error.
I have captured call with two agents on the line with a customers and
two
David @ULC ucoms2...@gmail.com writes:
Some at 5:34 pm EST DAILY, all my call get disconnect.
tcpdump. With a good trace, it should be fairly easy to figure out where
the problem is hiding.
Just be glad you have a specific time; comparing dumps isn't my
favourite pastime.
/Benny
Hi,
I've a few working asterisk servers, all seeing the same symptom, but
they are all based on the same configs.
A SIP inbound INVITE message is coming in to an extension (not a peer)
eg 5...@ourserver.com
A tcpdump clearly shows the INVITE coming in, but asterisk seems to be
Hello,
We have been working with the ReceiveFax application for some weeks now in
order to receive faxes in T.38 and it works fairly well, but there are some
faxes that for some reason we are not able to receive correctly.
The asterisk version we are using is 1.6.0.6 with spandsp-0.0.5pre4 and
Dear Users
Good day, need a help on connecting the FritzBox with my
Asterisk Server. Both are in LAN and from the Asterisk Server I can ping
the FritzBox. However the Username I gave in the box is somehow is not
geeting registered in the Asterisk application. The usetname I
configured in
Hi Santiago,
Santiago Gimeno wrote:
Hello,
We have been working with the ReceiveFax application for some weeks
now in order to receive faxes in T.38 and it works fairly well, but
there are some faxes that for some reason we are not able to receive
correctly.
The asterisk version we are
Hey there list !
I'm receiving negative feedback when people try to pickup another
ringing phone by pressing *8 on there own Grandstream device.
These are my setting that should make pickup possible :
all my sip-clients (Grandstream) have this in their config (sip.conf) :
callgroup=1
I only tried to connect my 7270 Fritz Box over a sip account on asterisk!
There are some points, you have to note:
- you have to select Using internet number
- in the area select other Provider
- in field internet number your asterisk number
- in field user name your number too
- your password
-
On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Days ago I bought a OpenVox A400P card with a port FXS and another FXO
that I am testing with my Asterisk installation in my house. I'm using
Asterisk 1.4.24.1 with DAHDI
On Wed, 20 May 2009, jonas kellens wrote:
Hey there list !
I'm receiving negative feedback when people try to pickup another
ringing phone by pressing *8 on there own Grandstream device.
These are my setting that should make pickup possible :
all my sip-clients (Grandstream) have this in
Hi All,
I am trying to implement ACD using Asterisk 1.2.18 and I've chosen
AgentCallbackLogin for login purpose. One AGI is written which will actually
get executed when agent dials '1001' (say) from his SIP phone and enters
into the queue. Second AGI gets executed when the Dial operation is
In article e77ab304d41c084090d498682873252fc1b...@nts-10.ca.hmhengineers.com,
Jimmy Ezell jez...@hmhca.com wrote:
Sounds like the workaround for 4.7 is to add this symlink that you mention.
What directory
does the symlink need to be in? What should it look like? Where should it
point
This all looks ok. What happens if you try to access the DAHDI channel
outside of Asterisk control:
In dialplan
Exten = 9,1,Dial(DAHDI/1)
Dial 9
Get dialtone
Dial number
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Hi,
Sorry for this newb question (but maybe a newb question once in
a while is ok):
What's the current state about Asterisk handling DTMF features via
SIP INFO (dtmfmode=info) even when the media path has been reinvited
(canreinvite=yes) to go directly from one phone to another?
Somewhat
Hi,
I'm getting the following error from an asterisk 1.6.0.9 installation:
[May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error:
Asked to delete sched id -1???
[May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: No
more room in scheduler
This repeats a few times,
This may or may not help, but put 'demo1' in ticks. Also, as I read this,
you're just testing extension 11?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Azher Mughal
Sent: Tuesday, May 19, 2009 10:07 PM
To:
Hello,
First of all I have an Asterisk setup of Asterisk 1.6.0.9 + DAHDI 2.0 +
E1 card with ISDN-15 line (KPN Netherlands).
I have two questions/situations:
A. I would like to be able to interrupt the dial command when I try to
call to a mobile phone and this phone is never answered by a person
Message: 19
Date: Tue, 19 May 2009 22:20:59 +0300
From: Tzafrir Cohen tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: asterisk-users@lists.digium.com
Message-ID: 20090519192059.gb3...@xorcom.com
Content-Type: text/plain; charset=us-ascii
Hi, I am running asterisk 1.6.0.5 with a Sangoma A104DE (4 port T1 with Echo
Cancellation).
We are using DAHDI.
When I do a top, I see asterisk using up 460MB of VM which is huge compared
to Asterisk system not using the card.
We also notice a constant decrease in available VM memory size on
Hi Steve,
Thanks for the answers.
Comments inline.
2009/5/20 Steve Underwood ste...@coppice.org:
Did you draw that arrow in the wrong direction? The side answering the
call should send the first V.21 signal.
No. That's what the wireshark trace shows.
The relevant information in the
Is it possible to specify the different pin for different user so that
I can identify who has joined the conference.
Jim
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asterisk-users mailing list
To UNSUBSCRIBE or update
On Wed, May 20, 2009 at 9:56 AM, Jim Boykin boykin...@gmail.com wrote:
Is it possible to specify the different pin for different user so that
I can identify who has joined the conference.
Not sure to your exact question, but I use the `i` option to MeetMe
MeetMe(100|I)
This has the callers
In article 4fbba87b0905200756s6e30e032g59aaf53f0ec09...@mail.gmail.com,
Jim Boykin boykin...@gmail.com wrote:
Is it possible to specify the different pin for different user so that
I can identify who has joined the conference.
I do it by using AGI to play a greeting and ask for a PIN. It then
Danny Nicholas schrieb:
This may or may not help, but put 'demo1' in ticks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Azher Mughal
my %out = $astman-sendcommand(
Hi,
On Wed, May 20, 2009 at 9:38 AM, Michel Verbraak mic...@verbraak.org wrote:
Is there an option for the dial command to stop the call when the switch is
detected and tell the caller that voicemail is active and if he would like
to leave a message or not? Can I create/detect this with an AGI
On Wednesday 20 May 2009 11:38:56 Azher Mughal wrote:
I have hint in extensions.conf as
exten = 30,2,hint,SIP/8172
exten = 30,hint,SIP/8172
--
Tilghman
___
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asterisk-users
BERGANZ François escribió:
Hello,
I have a var like 'blabla' with the '
I need to suppr the '
Is it possible with the ${var:x:y} ?
Thank you
It's like with any other string, so it should work.
Cheers,
Cordialement,
BERGANZ François
P Pensez à l'Environnement,
Hi,
I am doing something like this. Not actually detecting the switch, but
detecting vmail/answering machine. Take a look at this
http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect
Hope this helps.
_
From: asterisk-users-boun...@lists.digium.com
Cut should do this for you
Exten = x,x,Set(var2=cut(var1,\)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Wednesday, May 20, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject:
Dont work
I need that it suppr the
Thank you
Cordialement,
BERGANZ François
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : mercredi 20 mai 2009 17:53
À : 'Asterisk Users Mailing List -
I've got an Asterisk server, and several SIP phones behind our router
here. Things are working just perfectly inside the network, just as
the should.
However, I'm not trying to configure my asterisk server to talk with
SIP services outside our network, once such example is my gizmo
project
Santiago Gimeno wrote:
Hi Steve,
Thanks for the answers.
Comments inline.
2009/5/20 Steve Underwood ste...@coppice.org:
Did you draw that arrow in the wrong direction? The side answering the
call should send the first V.21 signal.
No. That's what the wireshark trace shows.
AFAIK, the pin is a conference specific feature. The I is a good idea but
not foolproof (user can record silence, etc). You might use the b option
to stream in an IP address when the user joins (This doesn't work with a SIP
channel according to the documentation) and have a lookup table that
Hello,
I have a var like blabla with the
I need to suppr the
Is it possible with the ${var:x:y} ?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
-- Bandwidth
1) it'll be hard to get 120 g729 calls with software codec unless you
have a super server with alot of logical CPU units ...
in that case it might be cost efficient to buy the transcoding card
2) you have to pay for the g729 codec licenses unless you want to use
it illegally
Martin
On Wed, May
I found !
exten = _X.,n,Set(var2=${CUT(var,',2)})
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ
François
Envoyé
Perhaps this
Exten = s,1,Set(myVar=123)
exten = s,2,Set(cutVar=${CUT(myVar|\|2)})
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Wednesday, May 20, 2009 11:21 AM
To: 'Asterisk Users Mailing List
On Tuesday 19 May 2009 22:04:06 Jason Aarons (US) wrote:
Windows Media Player thinks is a Intel G723 type 42 but can't play it..
This would explain why you cannot play it. G.723.1 is a codec that we
support for pass-through, but not for transcoding. Transcoding licenses
start at $325,000,
I have hint in extensions.conf as
exten = 30,2,hint,SIP/8172
but it throws error
WARNING[29338]: pbx.c:1833 pbx_extension_helper: No application
'hint' for extension
asterisk version is 1.4.25-rc1
Thanks
Philipp Kempgen wrote:
Azher Mughal schrieb:
I am trying to get the extension status
- Jonathan Moore supermegat...@gmail.com wrote:
I've got an Asterisk server, and several SIP phones behind our router
here. Things are working just perfectly inside the network, just as
the should.
However, I'm not trying to configure my asterisk server to talk with
SIP services
Azher Mughal schrieb:
I am trying to get the extension status (weather it has dialed
outgoing call via SIP or IAX2), using the following piece of code
however it always returns -1 on all the extensions (valid/invalid).
Just to be sure: Did you define hints for the extensions?
Philipp
On Wed, 20 May 2009, Deepak wrote:
Hi, I am running asterisk 1.6.0.5 with a Sangoma A104DE (4 port T1 with Echo
Cancellation).
We are using DAHDI.
When I do a top, I see asterisk using up 460MB of VM which is huge compared
to Asterisk system not using the card.
We also notice a constant
Have you tried relaxdtmf=yes in zapata.conf/dahdi.conf?
-Brent
Timm M.Schneider wrote:
Hi,
is there a possibility to tell zaptel or Asterisk to modify the DTMF
sensibility?
The problem what i have is that the Asterisk don't get all Numbers which the
analog-FAX dial, let say the FAX dial
Hi,
I am trying to pass DIALSTATUS to a Macro so that i can set a
variable when a call is placed (call is placed via a call file to
another extension first). Basically i don't want to dial a number
where a call is already bridged and thats why i am setting a variable.
[macro-afterdial];
exten =
Gordon Henderson schrieb:
I see a constantly growing memory footprint with asterisk 1.2 and zaptel
on analogue cards but with digium openvox TDM400 cards. I did ask about
it here some time back and got no reply - not even a we fixed this in
1.4 sort of reply, so these boxes get asterisk
On Wed, May 20, 2009 at 1:50 PM, Tim Nelson tnel...@rockbochs.com wrote:
Could you elaborate a bit more?
What isn't 'working out to well'?
Are you getting failed calls? One way or no audio?
Sorry for the lack of information. I posted in a bit of haste.
Initially it was failed calls, or not
Surprising that no one is experiencing/complaining. I'd assume a lot of
people out there would be running Sangoma T1 cards or other cards.
I contacted Sangome and they said it is an asterisk issue...
On Wed, May 20, 2009 at 12:54 PM, Gordon Henderson
gordon+aster...@drogon.net
No, you don't necessarily need a SIP proxy for this. Furthermore, while
a SIP proxy might assist with certain SIP-level reachability issues, it
will do nothing for the actual audio (media) if there are NAT issues
that prevent that from getting through.
As the other reply said, this isn't
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
What you have here should work just fine except:
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN}) -- note the change from n to 1.
I also don't understand why you have an Answer after your Dial statements.
I would
Azher Mughal schrieb:
I have hint in extensions.conf as
exten = 30,2,hint,SIP/8172
but it throws error
WARNING[29338]: pbx.c:1833 pbx_extension_helper: No application
'hint' for extension
That warning message tells you that you used hint as an application
by accident. hint is not an
Azher Mughal schrieb:
I am trying to pass DIALSTATUS to a Macro so that i can set a
variable when a call is placed (call is placed via a call file to
another extension first). Basically i don't want to dial a number
where a call is already bridged and thats why i am setting a variable.
Thanks.
Now when a call is connected i can see Idle shouldn't be 'In Use' :
*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
3...@demo: SIP/8172
State:IdleWatchers 0
- 1 hints registered
I have qualify=yes for all
Unfortunately, I don't have this phone and I can't find any
documentation for the 941 that refers to TLS setting. Here's what it
looks like when I set extension 4 to TLS on the 942:
SIP_Transport_4_ group=Ext_4/SIP_SettingsTLS/SIP_Transport_4_
Dimitris Counalakis wrote:
Thnx Mark,
but there
Azher Mughal schrieb:
Now when a call is connected i can see Idle shouldn't be 'In Use' :
*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
3...@demo: SIP/8172
State:IdleWatchers 0
- 1 hints registered
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Tony
Mountifield
Sent: Wednesday, May 20, 2009 06:26 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
In
What you say...Hose (hose+aster...@bluemaggottowel.com):
Hi,
I'm getting the following error from an asterisk 1.6.0.9 installation:
[May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error:
Asked to delete sched id -1???
[May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515
Thanks.
Philipp Kempgen wrote:
Azher Mughal schrieb:
Now when a call is connected i can see Idle shouldn't be 'In Use' :
*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
3...@demo: SIP/8172
State:IdleWatchers 0
Hi Listers,
I'm running 1.4.25-rc1 on opensuse 11.0 with
dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2.
Incoming calls work fine. Outgoing calls made directly (exten =
s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to
let
Hi,
I am attempting to make about ten calls simultaneously and intermittently
get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
busy/congested at this time
I am not sure if this is related to my bandwidth to my voip provider, a
configuration issue or something else.
Has anyone
On Wed, 20 May 2009, John Regal wrote:
Hi,
I am attempting to make about ten calls simultaneously and intermittently
get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
busy/congested at this time
I am not sure if this is related to my bandwidth to my voip provider, a
It might be related to bandwidth since each call takes 30-60kb depending on
codec. You could try putting a w in front of the number to make Dial wait
.5 seconds before starting.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
check if your dahdi card still takes interrupts at this point
dahdi_test should return some numbers close to 99%
Martin
On Wed, May 20, 2009 at 3:10 PM, Hose hose+aster...@bluemaggottowel.com wrote:
What you say...Hose (hose+aster...@bluemaggottowel.com):
Hi,
I'm getting the following error
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, May 20, 2009 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ...is circuit
Danny Nicholas wrote:
Hi Listers,
I'm running 1.4.25-rc1 on opensuse 11.0 with
dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2.
Incoming calls work fine. Outgoing calls made directly (exten =
s,1,Dial(DAHDI/G1) then number work fine. The problem I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tzafrir.
El miércoles 20 de mayo del 2009 a las 10:00:46 -0300,
Tzafrir Cohen escribió:
On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
Hint: you don't need to set 'signalling' for analog channels. Or just
set it explicitly
On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
multi-processor machine ( I had to remember to specify smp for the kernel)
I repeat: why bother with such an old system? Really?
Recall the comment from the book. That book had nothing really specific
to Centos 4. Why do you shoot
Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tzafrir.
El miércoles 20 de mayo del 2009 a las 10:00:46 -0300,
Tzafrir Cohen escribió:
On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
Hint: you don't need to set 'signalling' for analog
Using r/m because DAHDI takes 10-15 seconds to get TELCO rings.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List -
On Wed, 20 May 2009, John Regal wrote:
Thanks for the reply and apologize for the double post. My original post
landed in another thread and thought it may have been missed...
I questioned my voip provider before posting and they told me they have
other asterisk customers that are making
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
[snip]
I left the busy after dial because this is what the original poster
had. In this case, if the channel does not get hungup then the
On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
multi-processor machine ( I had to remember to specify smp
for the kernel)
I repeat: why bother with such an old system? Really?
Recall the comment from the book. That book had nothing really specific
to Centos 4. Why do you shoot
can anybody help me to give Opensource SIP client information which can be
modified as per our requirment
http://www.qutecom.org
---
Marek Cervenka
===
___
-- Bandwidth and
From the front page ( http://wiki.centos.org/FrontPage ):
*What is CentOS?*
CentOS is an Enterprise Linux distribution based on the freely available
sources from Red Hat Enterprise
Linuxftp://ftp.redhat.com/pub/redhat/linux/enterprise/.
Each CentOS version is supported for 7 years (by means of
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Dave.
El miércoles 20 de mayo del 2009 a las 18:12:04 -0300,
Dave Fullerton escribió:
I load the modules wctdm and dahdi. But when I execute in Asterisk
CLI dahdi show channels, I get the following error message:
No such command 'dahdi show
Many years in telecom and computer world is around 100 year in real life..
10 years ago i was a millionaire in the dot com boom and 24 years old with a
P2 300 computer.., 20 years ago i was military engineer and running on 3.76
MHz 386's amber screens.. last year it was dual cores, today its
I'm testing an upgrade of an i686 production machine running 1.4.24.1 to
1.6.0.9. I've installed dahdi-linux-2.1.0.4.
But:
asterisk -cvvv
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk
Is there a way to make the asterisk voicemail app play back messages in NEWEST
FIRST order, instead of OLDEST FIRST? I see the situation repeatedly where
someone needs to dip into their voicemail archive to get something from a
recently saved voicemail message, and they have to slog through
So you were fourteen and a military engineer?
j
On Wed, 20 May 2009, ContactTel Business wrote:
Many years in telecom and computer world is around 100 year in real life..
10 years ago i was a millionaire in the dot com boom and 24 years old with a
P2 300 computer.., 20 years ago i was
Hehe i meant 15 but i knew one would spot that..
I was 17 in fact, left at 22, yeah demolition, construction, sniper, road
demolish, anti tank craters, and all the bells and whistles,
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
I wrote a note earlier about this problem but have done quite a bit more
debugging. Now I'm stuck at what to do next.
I have inbound calls being answered by our Asterisk box, which then
dials our answering service and bridges those calls. The inbound and
outbound are both PRIs. The answering
Hi
Which is the best interface card to connect PSTN line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The Asterisk is in LAN and is reachable from all the IP phones in
the LAN.
Thanks
Manoj
On Tue, May 19, 2009 at 10:38:24AM +1000, Paul Hales wrote:
Not true. I am always wrong.
(wait...is that a paradox?)
only on the 42nd time
PaulH
[snip]
ContactTel Business wrote:
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Digium PSTN cards seem to work.
PaulH
Manoj Panicker - FOES wrote:
Hi
Which is the best interface card to connect* PSTN* line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The
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