AGX-Addons crashes Asterisk for us.
Working solution (on 100+ servers we installed):
-
apt-get -y install g++ libtiff4 libtiff4-dev patch autoconf automake
libtiff-tools
cd /usr/src
wget
http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20080402.tar.
Cool - thanks Rob. I will check it out tmorrow.
Simon
On Wed, Apr 16, 2008 at 4:34 PM, Rob Hillis [EMAIL PROTECTED] wrote:
IIRC Asterisk doesn't support the full presence publishing spec so you
won't get the full range of possible status types, however you should at
least get free/busy. I
Hello,
Let me ask for help on a problem
That I can not solve a
2 B410P on my server
I can not mounted ports TE UP
Everything seems to have been successfully compile no apparent errors
Misdn (1_1_7_2 version)
Zaptel (version 1.4.9.2)
Asterisk (version 1.4.18)
Kernel (version 2.6.17-5mdv)
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
Hi - Been using a TE205P for a number of months - no issues.
Today I was talking to someone and I heard click
No more phone service.
I still have data service on this T1 line. (partial phone)
zttool reports the SPAN as
Greetings list,
I've been playing around with queues on an old asterisk 1.2 box at a customer's
site. They want to be able to add really simple queue announcements every
minute, along the following lines:
sorry for the delay, someone will be with you shortly.
Looking at the announce options in
use the option periodic-announce
On Wed, Apr 16, 2008 at 1:47 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've been playing around with queues on an old asterisk 1.2 box at a
customer's site. They want to be able to add really simple queue
announcements every minute, along
Chris Bagnall wrote:
Greetings list,
I've been playing around with queues on an old asterisk 1.2 box at a
customer's site. They want to be able to add really simple queue
announcements every minute, along the following lines:
sorry for the delay, someone will be with you shortly.
I
Hi,
Johansson Olle E wrote:
Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by
checking HANGUPCAUSE instead of DIALSTATUS and you will get many more
details.
Great, that's all I need:
It gives me more ways to analyse the different reason for the hangup and
I can use the
Thanks, that`s what I ended up doing. Still, it doesn't seem to be WAD,
since the CDR(accountcode) is correct and suddently dissapears.
Is this a bug (I was looking through the bug system and couldnt match this
with a bug, but then again I am not a developer) or is it really WAD?
Mike
Hello Asterisk People,
I have two annoying bugs in asterisk, that i want to know if some of you
have already found a way to fix:
Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch.
1. I use a queue with just on sip device, one call at a time, however
and without reason
Heres something that's making me scratch my head... I'm using RxFAX on
ISDN lines and in-general it's going well.
However, there seems to be a case when the fax doesn't get delivered, but
looking through the CDRs it seems that the call happened, RxFAX was
executed .. time passed (1-2+
I forget another bug, i use the asterisk manager interface.
I frequently use the status function but it doesn't work as expected, i
use a program to parse the output of the status command but it don't
behave as expected, because i always wait for the latest package:
StatusComplete, and this
I'm trying to get the Telco MWI recognition working in asterisk-1.6-beta7. I'm
told that it's supposed to work provided
my telco support FSK MWI signalling.
I have Verzon FIOS. I believe I have FSK MWI signaling as I can hear the
standard stutter tone when I pick up a live
handset in front
We have two servers but looks like G729 issues. Works fine on the old server
and not sure if it is T1 related. See SIP Debug. Any experiences to share.
Thanks
---
Newark1*CLI
--- SIP read from 194.xx.Xx.Xx:5060 ---
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP
On Wed, 2008-04-16 at 07:55 -0500, Nestor A. Diaz wrote:
I frequently use the status function but it doesn't work as expected, i
use a program to parse the output of the status command but it don't
behave as expected, because i always wait for the latest package:
StatusComplete, and this
On Tue, Apr 15, 2008 at 7:07 PM, Shaun Ruffell [EMAIL PROTECTED] wrote:
Your stack trace appears to possibly be stack corruption.
Could you try either this branch:
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Just tried it... Behaviour looks equivalent. Drivers
Hello, all!
I've noticed a peculiar situation and I am hoping someone can shed
some light on it for me. We have an Asterisk (1.4.18 ) box talking to
the world via Zaptel on a PRI from a telco (USA). I have an extension
that returns busy signal (fast-busy or regular busy) (using US tones).
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]mailto:SIP/[EMAIL PROTECTED]
How can you use
Could be this...
http://www.misdn.org/index.php/FAQ_chan_mISDN#Why_does_the_L1_goes_DOWN_on_my_PMP_Isdn_Link.3F_Or_why_do_i_get_No_free_chan_even_after_group_call_from_chan_misdn_if_dialing_out_on_my_PMP_Link.3F
Hmmm... that's a long link. It is the
Why does the L1 goes DOWN on my
Asterisk Users,
I am running a Debian Etch system with Asterisk 1.4.11 with a TDM03B card.
Once in awhile, I get this error on the Asterisk, which causes my channels to
be busy/congested, leaving me with just one channel to recieve and make calls:
NOTICE[31454]: chan_zap.c:6367 ss_thread:
Ex Vito wrote:
Hi list,
After a lot of testing + troubleshooting, I guess I'm observing
what I am now calling a regression with zaptel 1.4.10 (is it?)
As such I call for peer feedback, before either asking Digium
install support or filing a bug.
Thanks in advance!
Hi Raul,
CDR's for transfers are beyond the ability of Asterisk.
http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html
http://bugs.digium.com/view.php?id=11093
It's not something the powers that be want to think about a design for
and the solution that's been suggested is to
Doug Lytle wrote:
Chris Bagnall wrote:
Greetings list,
I've been playing around with queues on an old asterisk 1.2 box at a
customer's site. They want to be able to add really simple queue
announcements every minute, along the following lines:
sorry for the delay, someone will be with
On Wed, 16 Apr 2008 08:40:42 -0500, Mark Gimelfarb [EMAIL PROTECTED]
wrote:
why do cell phones and Gizmo both detect busy tones and terminate the
call? Is that a standard behavior?
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number
What country are you in?? Yes, it is common for cell phones to
disconnect the call if they receive CONGESTION, but not BUSY.
Horwich IT Services (Godwin Stewart) wrote:
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number is busy. It
Hi, I need to make Click-to-Call web application to connect with an asterisk
server.
I´m using Java
What solution recommend me?
Thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850
or 2650 (cannot recall):
00:00.0 Host bridge: Broadcom CMIC-WS
I'm in the US, so I was originally using the US tones.
Looks like I'm getting a disconnect with both CONGESTION and BUSY. In
fact, I wasn't actually using Congestion() and Busy(), I just did
Playtones() for both of those. There is no reason to send PRI messages
to cell phones, is there? The
Drew Gibson wrote:
Works fine for me on 1.2.24 ...
Sorry, I thought periodic announcements were a 1.4 thing.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote:
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed in a Dell PowerEdge
Ex Vito wrote:
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
The softlockup indicator should be benign. It gets called when loaded
the firmware for the part since the firmware image is so large and it
takes a long time to load. However, I might have a fix
Hi Al,
Al Baker wrote:
Shaun - Could you clarify your post a bit ?
1 - Is the 4 K stacks a Known Problem ?
a) If so is it known to be problem on any specific Linux distro ?
b) Should ALL installation Check for this PRIOR to doing an
Asterisk Install ?
I wouldn't really say a
Shaun Ruffell wrote:
Hi Al,
Al Baker wrote:
Shaun - Could you clarify your post a bit ?
1 - Is the 4 K stacks a Known Problem ?
a) If so is it known to be problem on any specific Linux distro ?
b) Should ALL installation Check for this PRIOR to doing an
Asterisk Install ?
Re,
That is correct, in the case of France Telecom
Pb resolved.
With my thanks
A +
Yves
Le mercredi 16 avril 2008 à 14:45 +0100, Ex Vito a écrit :
Could be this...
Tzafrir Cohen wrote:
On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote:
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
The softlockup indicator should be benign. It gets called when loaded
the firmware for the part since the firmware image is so large and it
takes a long time to load. However, I might have a fix for you.
Can
equis software wrote:
Hi, I need to make Click-to-Call web application to connect with an
asterisk server.
I´m using Java
What solution recommend me?
I did a spiel on this at Astricon last year. The slide deck is
somewhere around for those interested, but now we also have some code to
On Wed, Apr 16, 2008 at 04:11:52PM +0100, Ex Vito wrote:
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
[snip]
Can you try my stack reduction branch at:
https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup
If that does not work,
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
One thing also I would like to see is your kernel .config file. Another
thing that would for sure remove that warning is to disable the kernel
softlockup detector which is giving a false lockup warning in this
Ex Vito wrote:
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
One thing also I would like to see is your kernel .config file. Another
thing that would for sure remove that warning is to disable the kernel
softlockup detector which is giving a false lockup
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Question:
- The url you suggest is very similar, are we talking about
a different stackcleanup branch ?
Try:
http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Try the
On Wed, Apr 16, 2008 at 4:46 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
It's the same. Sorry, I sent you that email before I saw his message.
I just got an idea for a clever way to make the softlockup detector not
complain. I'll let you know when I have a patch to try.
...sure.
Well, I think this should be a critical feature to implement for the next
releases of Asterisk (hopefully * 1.6), I've read a lot of this matter in
the list and in the bug tracker. I think the more insightful reading about
this topic can be found in this link: http://www.asterisk.org/node/48358
I
Nestor A. Diaz wrote:
1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip show inuse and this is the result:asterisk
Anonymous wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback('
Check the web embedded click-to-call solution from videoreps.net. It is
free. It includes click-to-video, click-to-call, and click-to-did
CS
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of equis software
Sent: Wednesday, April 16, 2008 7:36 AM
To: Asterisk Users
update with no 4K stack kernel:
- The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5
- The only .config change was to disable the CONFIG_4KSTACKS
Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as
suggested by Shaun and Mathew.
Short: Results are about
Well,
Installed asterisk, libpri, zaptel,... trunk
Parameters seems ok for asterisk and ss7, linkset is ok
Problem is astersik doesn't matter about the sip messages sent to him,
Ngrep see the messages on port 5060 but astersik doesn't react...
Even sip set debug on doesn't give me any infos...
Well,
Installed asterisk, libpri, zaptel,... trunk
Parameters seems ok for asterisk and ss7, linkset is ok
Problem is astersik doesn't matter about the sip messages sent to him,
Ngrep see the messages on port 5060 but astersik doesn't react...
Even sip set debug on doesn't give me any infos...
Dear all,
A quick question on deploying Asterisk over E1. I am looking for a low-cost
solution for bridging my E1 line and Asterisk with reasonable stability
suppoing both voice and fax. Will a Digium T100 be good for that or I
really need a Cisco AS 5400 for this task? What is the difference
Mojo with Horan Company, LLC wrote:
Nestor A. Diaz wrote:
1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip
Ex Vito wrote:
update with no 4K stack kernel:
- The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5
- The only .config change was to disable the CONFIG_4KSTACKS
Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as
suggested by Shaun and Mathew.
Hi all,
I've just upgraded to 1.4.19 from 1.4.18.1 and now have problems with
app_chanspy. To monitor I use -
exten = 596,1,ringing
exten = 596,n,Wait(1)
exten = 596,n,ChanSpy(|g(2000))
exten = 596,n,Hangup
and the listened-to channel as follows -
exten = _77,1,Set(SPYGROUP=2000)
exten =
Hi,
I`m trying to use Chanspy for a customer that wants to listen to his
employees so he can train them better (or so he claims). In any case, it
looks simple but there is something I`m not doing right.
When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234)
When I use, on another
If your requirements are simple and you only have a small number if E1s,
you can also use a Cisco 36xx with a T1/PRI card. 3600's have limited
capacity but we run 4 PRIs on a 3640 no problem and it's been very stable
for several years. The nice thing about 3600's is they are almost free,
--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Vieri wrote:
How can I tell the make system in 1.4.19 that ilbc
is
already on the system and that it should link to
/usr/lib/libilbc.a?
Shouldn't the configure script do that?
No; the Asterisk build system has never had support
for
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
One thing you can also do is pass the nosoftlockup kernel parameter
into the kernel from the bootloader. That should disable the softlockup
detector.
Tested with no 4K stack kernel and stackcleanup svn branch
Try using chanspy without setting the variable first. This should give you a
broader base of channels. Then start to narrow it down.
On Wed, Apr 16, 2008 at 8:33 PM, Mike [EMAIL PROTECTED] wrote:
Hi,
I`m trying to use Chanspy for a customer that wants to listen to his
employees so he can
Ex Vito wrote:
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
One thing you can also do is pass the nosoftlockup kernel parameter
into the kernel from the bootloader. That should disable the softlockup
detector.
Tested with no 4K stack kernel and
Hi,
(me again, my upgrade to 1.4 is more painful then I imagined it would be).
I just noticed that the command show hints shows all hints correctly, but
none of them ever are InUse (even if I use a line and dial out) like I used
to on 1.2.
Can`t find a bug in the bug tracking system, is
Hi list,
Any good drag and drop transfer call application for windows based systems
you can advise ?
Something like HUD perhaps?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
Mojo with Horan Company, LLC wrote:
Nestor A. Diaz wrote:
1. I use a queue with just on sip device, one
call at a time, however
and without reason just after some couple of
hours the sip device show
in use and then no calls are
Mojo with Horan Company, LLC wrote:
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
Oops,
shouldn't have that second priority in there. Because Read is playing
Mike wrote:
Hi,
I`m trying to use Chanspy for a customer that wants to listen to his
employees so he can train them better (or so he claims). In any case,
it looks simple but there is something I`m not doing right.
When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234)
|=, as of 1.2 IIRC
On Wed, Apr 16, 2008 at 9:54 PM, Thomas Kenyon [EMAIL PROTECTED]
wrote:
Mike wrote:
Hi,
I`m trying to use Chanspy for a customer that wants to listen to his
employees so he can train them better (or so he claims). In any case,
it looks simple but there is something
I am getting an error:
chan_zap invalid tone duration 11220.
This is line 11220 in chan_zap.c and I have a toneduration of 300 in the
zapata.conf file.
I have commented it out and it is now working again.
Why is that an invalid paramter? It never used to be.
jerry
Al lists wrote:
Hi list,
Any good drag and drop transfer call application for windows based
systems you can advise ?
Flash Operator Panel (FOP)
http://www.asternic.org
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
On Wed, Apr 16, 2008 at 03:51:02PM -0400, Jerry Geis wrote:
I am getting an error:
chan_zap invalid tone duration 11220.
This is line 11220 in chan_zap.c and I have a toneduration of 300 in the
zapata.conf file.
I have commented it out and it is now working again.
Why is that an
Al lists wrote:
Hi list,
Any good drag and drop transfer call application for windows based
systems you can advise ?
Something like HUD perhaps?
Yes.
Maestro Control Panel (I authored this one)
http://www.datatrakpos.com/pos/datatalk/maestro.aspx.
There is also the nice flash based
Introducing Click-to-Call
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media that
a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support etc have asked for
a Click-to-Call service.
On Wed, 2008-04-16 at 18:51 +0100, Steve Rawlings wrote:
This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered
but there's no spying, the only way I could get this to work was with -
exten = 596,n,ChanSpy(|b)
but this spied on all channels, not just those with SPYGROUP
On Wed, 2008-04-16 at 14:20 -0400, Mike wrote:
I just noticed that the command show hints shows all hints
correctly, but none of them ever are InUse (even if I use a line and
dial out) like I used to on 1.2.
Can`t find a bug in the bug tracking system, is there something else I
should
Rhino or audiocode PSTN gateway
- Original Message -
From: mark morreny
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PSTN to SIP
Date: Thu, 17 Apr 2008 01:25:49 +0800
Dear all, A quick question on deploying Asterisk over E1. I am
Introducing Click-to-Call
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media that
a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support etc have asked for
a Click-to-Call service.
Lee Jenkins wrote:
Al lists wrote:
Hi list,
Any good drag and drop transfer call application for windows based
systems you can advise ?
Something like HUD perhaps?
Yes.
Maestro Control Panel (I authored this one)
http://www.datatrakpos.com/pos/datatalk/maestro.aspx.
There is also
On Wed, Apr 16, 2008 at 9:10 AM, broadband Voice
[EMAIL PROTECTED] wrote:
We have two servers but looks like G729 issues. Works fine on the old server
and not sure if it is T1 related. See SIP Debug. Any experiences to share.
Thanks
---
Newark1*CLI
--- SIP read from 194.xx.Xx.Xx:5060 ---
Bob G wrote:
Introducing Click-to-Call http://1ezphone.com/
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media
that a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support
I want to add to my dialplan the ability to spy on an arbitrary
extension whether a call originates at it or is terminated at it.
Scenario 1: Given an extension, say 2001, a call comes in on a zap
channel and is Dial()ed to the phone that's at extension 2001, I want to
be able to pick up a phone
Why the guy asked a question?
- Original Message -
From: Lee Jenkins
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Drag and Drop transfer application
Date: Wed, 16 Apr 2008 16:21:54 -0400
Bob G wrote:
Introducing Click-to-Call
Vieri wrote:
So basically I'm wondering if the Asterisk
make/configure process could do steps 1 and 2
automagically for me.
I can't find any other Linux distribution that provides libilbc, so this
would be a very Gentoo-specific change if we did it. Also, we'll have
the iLBC source code back
On Wed, 16 Apr 2008, Bob G wrote:
Introducing Click-to-Call
So, since you posted this on a non-commercial discussion list, this is
available for free?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice:
On Wed, 16 Apr 2008, Bob G wrote:
Why the guy asked a question?
From: Lee Jenkins
Bob G wrote:
Introducing Click-to-Call
I think you're going to get yelled at ;)
1) You hijacked the thread.
2) You top-posted.
3) It's a non-commercial list -- RTFMLIBP (Read the Mailing List
Brian J. Murrell wrote:
Does anyone have an implementation of this they'd like to share?
I cut out the authentication stuff we do, but this is part of the macro
we use to spy and record calls arbitrary calls. All of our users have
sip handsets. Asterisk 1.2.
exten =
Jerry Geis wrote:
I am getting an error:
chan_zap invalid tone duration 11220.
Is this actually the error message you got? I don't see the line number
being placed into the error message by the code in chan_zap. When
reporting errors, it is very helpful if you actually copy and paste the
Steve Edwards wrote:
On Wed, 16 Apr 2008, Bob G wrote:
Why the guy asked a question?
From: Lee Jenkins
Bob G wrote:
Introducing Click-to-Call
I think you're going to get yelled at ;)
1) You hijacked the thread.
2) You top-posted.
3) It's a non-commercial list --
Bob G wrote:
Why the guy asked a question?
Yes. But the question was about Drag and Drop transfer applications for
Asterisk.
Can 1EZphone do that? If not, your SPAMMING the list!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Wed, Apr 16, 2008 at 03:24:15PM -0500, Bob G wrote:
Why the guy asked a question?
And you did not provide a useful answer to it. You merely quoted a
leangthy press release. It might have been partially relevant. And might
not.
Stick to relevant answers. Certainly so when promoting your
On Wed, 2008-04-16 at 13:47 -0700, Steven Kurylo wrote:
exten = s,n(getext),Read(SPY,extension,4)
exten = s,n,GotoIf($[ ${LEN(${SPY})} != 4 ]?nospy)
exten = s,n(spy),UserEvent(ChanSpy,User ${CALLBACKNUM} spied on ${SPY})
exten = s,n,Chanspy(SIP/${SPY},r(monitor-ext-${SPY}))
exten =
Hi There,
We have our Asterisk box using a external SIP provider for outgoing
calls over our DSL line. This seems to be going well... But i do have
the ability to set some QOS ports in our linksystem DSL router... Its
faily basic, so im wondering if it will help at all...
We can specify High,
Sorry
- Original Message -
From: Tzafrir Cohen
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Drag and Drop transfer application
Date: Thu, 17 Apr 2008 00:00:57 +0300
On Wed, Apr 16, 2008 at 03:24:15PM -0500, Bob G wrote:
Why the
I have just noticed the L() argument to Dial() and it seems pretty
obvious that this could be used to create a lightweight prepaid calling
system.
I'm wondering if anyone has some extensions.conf dialplan using
Dial(..., L(...)) and the astdb to do lightweight prepaid service. I
only need to
Has anyone used or thought of using Asterisk server farm behind LVS.
-Jai
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Gordon Henderson wrote:
Heres something that's making me scratch my head... I'm using RxFAX on
ISDN lines and in-general it's going well.
However, there seems to be a case when the fax doesn't get delivered, but
looking through the CDRs it seems that the call happened, RxFAX was
executed
Hi,
Our requirement is just to be able to do voice and fax at a quality manner.
What is the difference between using a physical server vs a PCI card that
plugs in
to the Asterisk server? Is there a big difference in terms of scalability?
We are looking at a solution that can be easy-to-deploy
On Wed, Apr 16, 2008 at 11:49 PM, Simon [EMAIL PROTECTED] wrote:
Hi There,
We have our Asterisk box using a external SIP provider for outgoing
calls over our DSL line. This seems to be going well... But i do have
the ability to set some QOS ports in our linksystem DSL router... Its
faily
The Cisco's also support T.38 gateway functions whereas Asterisk can
only do pass thru. Either way you'll still need another server,
typically hylafax, to receive the faxes to get them somewhere useful.
In my experience the Cisco switches are definitely the way to go for
the ISDN/SIP gateway and
hi
i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 ,
0285469658 etc.
is it possible to keep users from using the 0282549087 line always open that
it only allows a certain user to make outgoing calls on it?
-
Be a
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