Steve, I downloaded the latest Asterisk version (see below).
*CLI core show version
Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on
2008-09-11 06:10:06 UTC
If I code:
Hint(Custom:light1)
It will pass aelparse but when it runs, it says Hint is an unknown
application on
Hello List,
first please excuse my not so perfect english, since it is not my native
language.
I have problem concerning queues and connected services.
The scenario:
I define agents in an agent.conf file using chan_agent and these
agents are made members of a queue in the queues.conf.
An
Greetings!
I have got a systematic problem with the SHELL function. Consider this
dialplan snippet:
*** CUT ***
exten = NUM,1,Set(myreturn=${SHELL(ast_picker sound_file)})
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${myreturn} = 0]?4:6)
[...]
*** CUT ***
ast_picker does simultaneously
On Thu, 2008-09-11 at 17:41 +1000, Lee, John (Sydney) wrote:
Steve, I downloaded the latest Asterisk version (see below).
*CLI core show version
Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on
2008-09-11 06:10:06 UTC
If I code:
Hint(Custom:light1)
It will pass
2008/9/11 CunningPike [EMAIL PROTECTED]
Oliver,
We use DNS SRV records combined with short TTLs
How short ?
to provide failover.
Thankfully, we have only used it when moving phones from one server to
another in preparation for upgrades, but it worked like a champ then.
I was first
Hi,
Gordon Henderson schrieb:
On Thu, 4 Sep 2008, Tharanga wrote:
Hi folks,
Can some one recommend a good video phone for asterisk (SIP.IAX2). I need
better quality, duarability. and should support various video codec's
.(Codec auto negotiation support id prefferble)
[...]
Some
Dear,
I have some limitations to install zaptel because of kernel reinstalling.
also there is'nt any zaptel device installed in the server.
but I need to install meetme, for conferencing .
can u help me ?
Best
Mani
___
-- Bandwidth and
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
winmail.dat___
-- Bandwidth and Colocation Provided by
Greetings list,
We recently installed some Linksys SPA962 + SPA932 sidecars into a client's
offices. The BLF functionality works fine, but call pickup using the BLF is
something of an issue.
Following the advice on voip-info.org, I configured part of their dialplan as
follows:
exten =
You only need to install ztdummy.
It's usually straightforward if you have Linux kernel 2.6.
-quan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pezhman Lali
Sent: September-11-08 5:59 AM
To: asterisk
Subject: [asterisk-users] meetme without zaptel
Dear,
I have some limitations
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this
particular application.
If the users in question are often in hotels abroad, something like this may
not solve the problem - I've noticed quite a few hotels are now blocking SIP
traffic (presumably so as to
Chris Bagnall wrote:
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this
particular application.
If the users in question are often in hotels abroad, something like this may
not solve the problem - I've noticed quite a few hotels are now blocking SIP
thanks for reply
is ztdummy installed Independence of kernel,?
--- On Thu, 9/11/08, Quan Nguyen, (NCS) [EMAIL PROTECTED] wrote:
From: Quan Nguyen, (NCS) [EMAIL PROTECTED]
Subject: RE: [asterisk-users] meetme without zaptel
To: [EMAIL PROTECTED] [EMAIL PROTECTED], Asterisk Users Mailing List -
On Thu, Sep 11, 2008 at 02:58:59AM -0700, Pezhman Lali wrote:
Dear,
I have some limitations to install zaptel because of kernel reinstalling.
Zaptel does not require completely reinstalling the kernel. But it does
require loading an extra kernel module.
also there is'nt any zaptel device
Chris Bagnall schrieb:
snip
Hello,
first you have to use the lastest firmware for the spa962. When you have
this installed you will see a input field for pickup code in the webif
for the spa932
just put a # after the pickup code you want to use (**# should work for you)
the # after the pickup
Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
I just whipped this up to test and it works for me in
Hi;
I want to built n-way-call-conference like this:
I call user 1, when he answers, I redirect him in one conference room
I call user 2 , when he answers, I redirect him in the same conference room
I do the same thinks with all users.
I don't know how I can do that?
Some one have any
Hello!Please join my Professional Network on iDeezire.To join, you must first register on the website by clicking the link below: Click here to join.Sincerely,Sarfarazhttp://www.ideezire.com - This invitation was sent by iDeezire on behalf of Sarfaraz.
Now that we have voicemail working, people have asked to be able to
dial in externally and be able to access their voicemail. My dial plan is
simple, after ringing a few extensions for some time, it goes to voicemail.
What needs to happen to allow for someone to switch out of this into
Yo can do it with Playtones(!440) !440 is for france seach yours in
indications.confhere is the example script from asterisk-france, the guy had
the exact same problem
[Appel_Sortant_Isdn]
exten = _0,1,Set(Flag_Playtone = 0)
exten = _0,n,Playtones(!440)
exten = _0,n(Continue),Read(Digits,,1,,,3)
Dear all,
I've used trixbox to compose a custom IVR, and I've defined the input
of pound key(#) leading to the repetition of the parent announcement.
But each time the pound key is pressed, file dir-intro.gsm will
always be played. Can any one tell me the reason?
By the way,if I want to modify
Press *
On 11 Sep 2008, at 14:31, Joseph L. Casale wrote:
Now that we have voicemail working, people have asked to be able to
dial in externally and be able to access their voicemail. My dial
plan is
simple, after ringing a few extensions for some time, it goes to
voicemail.
What needs
On Thursday 11 September 2008 03:09:19 Julien Claassen wrote:
Greetings!
I have got a systematic problem with the SHELL function. Consider this
dialplan snippet:
*** CUT ***
exten = NUM,1,Set(myreturn=${SHELL(ast_picker sound_file)})
exten = NUM,2,Answer()
exten =
[EMAIL PROTECTED] wrote:
Is the idea to switch to another video source or stay with the callers
camera? An option for both would be nice. I could see a help desk
placing a caller in que and a 1-2 min video coming on showing some
simple video of how to hook it up.
What I had in mind
Dear Max,
Yes, I have modified line no 5 with include_once(dirname(__FILE__) .
/lib/fpdf.php');
But I can not also see export to csv or export to pdf option from
download cdr data as I required within date.
--
With Regards,
Hiren Mistry
___
--
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I
try to make menuseletc I get the following error.
This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running
an up to date Debian etch.
Asterisk builds okay (not tried running it yet)
menuselect_gtk.c: In
It sounds like you probably need to install libgtk2.0-dev, or whatever
the package is, that contains the GTK headers needed for compilation.
On Thu, September 11, 2008 10:57 am, Thomas Kenyon wrote:
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I
try to make menuseletc
2008/9/11 Stefan Schmidt [EMAIL PROTECTED]:
Chris Bagnall schrieb:
snip
Hello,
first you have to use the lastest firmware for the spa962. When you have
this installed you will see a input field for pickup code in the webif
for the spa932
just put a # after the pickup code you want to
Hello,
I'm doing g729 passthrough with asterisk 1.4 and it is working great whene i
call directly from my softphone to the destination number, but i'm not able
to do passthrough whene i make calls via Manager API Originate Command, the
calls always fail
Hi. I am using asterisk 1.4 branch svn from yesterday and I am having
some problems -- I am still using the zaptel drivers temporarily.
Now there are two problems -- the first minor in that my asterisk
would not work at all when I first tried it -- or at least not on my
X400p card chan_dahdi
Hi!
Thanks! I changed course and reworte the program code to interoperate in
other ways. Now it works.
Is there a way to do something based on some other phone taking a call? Or
if the caller stops ringing?
It's too bad asterisk can't run applications/functions in parallel...
Kindest
Hi All,
I'm trying to incorporate callerid_shell.agi into my Asterisk 1.4.21.2.
I'm having trouble with it only returning data from the nanpa database. If
I fire it up manually, I get the correct data from the sqlite3 database.
What is everybody using for callerid resolution for their
On 9 Sep 2008, at 20:19, Mattias Andersson wrote:
Hi all!
I am looking for some software that would work as a proxy between a
SIP device (SIP phones and ATA boxes) and the Asterisk system
running IAX. The reason is that I can only get IAX trow the
firewalls, and can't change the
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok.
Just found
Did you setup the new /etc/dadhi/system.conf as well as unloading your old
zaptel modules and re-inserting the new dahdi modules?
* The primary kernel modules have changed names; the new names are:
zaptel.ko-dahdi.ko
ztd-eth.ko - dahdi_dynamic_eth.ko
On Thu, 11 Sep 2008, Julien Claassen wrote:
It's too bad asterisk can't run applications/functions in parallel...
You can write a multi-threaded application or a multi-threaded AGI.
Thanks in advance,
Steve Edwards
Steve Davies schrieb:
Thanks for that excellent information - Now does anybody know the XML
to provision that field? Normally you take the text on the screen
Call Pickup Code and replace space with underscore
Call_Pickup_Code ua=na *8# /Call_Pickup_Code
Unfortunately Call Pickup Code
Hmm... Ok, I tried that but encode.h and decode.h were not found...
On Wed, Sep 10, 2008 at 11:23 AM, Thomas Kenyon
[EMAIL PROTECTED] wrote:
Edgar Guadamuz wrote:
I notice that I have only format_ilbc.so but not codec_ilbc.so... is
it due to the compilation or there is some way to create
I was still using the zaptel kernel drivers -- this is what I would
like to do for now.
on Thursday 09/11/2008 Matt Gibson([EMAIL PROTECTED]) wrote
Did you setup the new /etc/dadhi/system.conf as well as unloading your old
zaptel modules and re-inserting the new dahdi modules?
* The
Done!
I had to do
# cd /usr/src/asterisk
# ./contrib/scripts/get_ilbc.sh
and the compilation was successful
On Thu, Sep 11, 2008 at 11:55 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
Hmm... Ok, I tried that but encode.h and decode.h were not found...
On Wed, Sep 10, 2008 at 11:23 AM,
On Thu, 11 Sep 2008, Russell Bryant wrote:
[EMAIL PROTECTED] wrote:
Is the idea to switch to another video source or stay with the callers
camera? An option for both would be nice. I could see a help desk
placing a caller in que and a 1-2 min video coming on showing some
simple video of
On Thu, Sep 11, 2008 at 12:41:12PM -0400, Jay R. Ashworth wrote:
Will I actually need to do PRI debug on that span to tell?
I did a pri debug to a file, I can see the call go, I see no indication
that it actually tried to generate a TBCT/RLT request.
Cheers,
-- jra
--
Jay R. Ashworth
-Original Message-
From: John covici [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 11, 2008 1:52 PM
To: Matt Gibson
Cc: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [asterisk-users] dahdi vs zap with latest version of asterisk
-- having
Hi!
AGI... Oh I was never good at perl. I'mhappy, that I finally completed my
personal ringing application as a standalone program.
Though I'd be only too willing to share the code, if someone wants/could
make an asterisk application of it.
I think I may still just bore down on AGI and
Greetings!
Does application Jack run the whole time, the conversation is going?
If so: is there a SIMPLE extensions.conf-only-based way to put it in the
background? I know AGI and other applications... :-(
Kindest regards and thanks
Julien
Music was my first love
On Thu, Sep 11, 2008 at 01:52:02PM -0400, John covici wrote:
I was still using the zaptel kernel drivers -- this is what I would
like to do for now.
In Asterisk 1.4: you can.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
Hey hey...
I'd like to create a new feature code in asterisk so when a user dials...
say... *00, it would then call some other extensions and play a sound file
to them.
So far, this is what I have...
[testing-custom]
exten = *00,1,Wait(1)
exten = *00,2,Playback(beep)
exten =
Hi All,
I'm trying to incorporate callerid_shell.agi into my Asterisk 1.4.21.2.
I'm having trouble with it only returning data from the nanpa database. If
I fire it up manually, I get the correct data from the sqlite3 database.
What is everybody using for callerid resolution for their
Hi all,
I am unable to run make menuselect for asterisk-addons. Works fine for zaptel
and asterisk. Here is the output.
Jonn
[EMAIL PROTECTED] asterisk-addons]# make menuselect
CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent makeopts
\make[1]:
Asterisk 1.6.0rc6 and Asterisk-addons 1.6.0rc1
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor
Sent: Thursday, September 11, 2008 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Unable to run make menuselect for
Jonn R Taylor wrote:
Hi all,
I am unable to run make menuselect for asterisk-addons. Works fine for
zaptel and asterisk. Here is the output.
Jonn
[EMAIL PROTECTED] asterisk-addons]# make menuselect
CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
Personally, instead of doing all that, I would simply add _one_
provider, then
when a call comes in on one DID, send it to one context/IVR, and when
the other
call comes in on the other DID, send it to it's own context/IVR.
I have done setups like this for bandwidth with 40+ DID's, you only
On Sep 10, 2008, at 7:11 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote:
On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote:
Hi to all, I actually have an asterisk server configured to
On Thu, 11 Sep 2008, Julien Claassen wrote:
AGI... Oh I was never good at perl. I'mhappy, that I finally completed my
AGI does not imply Perl. While I prefer to write AGIs in C, I have also
used PHP and Bash.
Thanks in advance,
On Thursday 11 September 2008 15:47:19 Eric Chamberlain wrote:
On Sep 10, 2008, at 7:11 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote:
On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote:
On Wednesday 10 September 2008 13:22:51 Ricardo Melendez
On Thu, Sep 11, 2008 at 03:18:38PM -0500, Jonn R Taylor wrote:
Hi all,
I am unable to run make menuselect for asterisk-addons. Works fine for zaptel
and asterisk. Here is the output.
Am I the only one who finds it strange that zaptel and asterisk-addons
check for gtk? Aren't we just a
At 2:29 PM on 11 Sep 2008, Mike Johnson wrote:
Hey hey...
I'd like to create a new feature code in asterisk so when a user
dials... say... *00, it would then call some other extensions and
play a sound file to them.
So far, this is what I have...
[testing-custom]
exten =
Jonn R Taylor wrote:
I am unable to run make menuselect for asterisk-addons. Works fine for
zaptel and asterisk. Here is the output.
Would you mind opening an issue on mantis (http://bugs.digium.com/) and
attaching both your asterisk config.log and the menuselect config.log?
Jonn R Taylor wrote:
I am unable to run make menuselect for asterisk-addons. Works fine for
zaptel and asterisk. Here is the output.
Sorry, your asterisk-addons (not asterisk) and menuselect config.log files.
Thanks,
--
Sean Bright
[EMAIL PROTECTED]
I have an Asterisk box at a clients site, and they are having quality
issues, they are using g711 for the codec and I think it might be that
asterisk is just overloaded.
Upstream I did a packet capture. The capture showed traffic going to
the asterisk box from the ISP is clean, but traffic from
At 8:29 AM on 11 Sep 2008, John Millican wrote:
Not directly on-topic for this list, but I'd not heard of OpenSIPS
before, so I had a look at the website. It looks to be a fork of
OpenSER. Does that mean OpenSER development has slowed/ceased, or
has the OpenSER project itself morphed
On Sep 11, 2008, at 2:18 PM, Julien Claassen wrote:
Does application Jack run the whole time, the conversation is going?
If so: is there a SIMPLE extensions.conf-only-based way to put it
in the
background? I know AGI and other applications... :-(
Yes. When you use the Jack
From memory, this is an issue with Asterisk 1.2 which can be fixed by
moving to 1.4
PaulH
Chris Bagnall wrote:
Greetings list,
We recently installed some Linksys SPA962 + SPA932 sidecars into a client's
offices. The BLF functionality works fine, but call pickup using the BLF is
hi sir,
thank you for your reply, i actually had the setup that way, but then
came a problem of limiting a number of calls for each customer.
i tried using setgroup to count number of calls, but my asterisk is on a
cluster, so counting of number of groups in asteriskA is different from
Thomas Kenyon wrote:
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I
try to make menuseletc I get the following error.
This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running
an up to date Debian etch.
Asterisk builds okay (not tried running it
Press *
Steven,
Appreciate the info but there must be something I missing as a
prerequisite to this feature. It has no effect at any point during the
call and message?
Thanks!
jlc
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
I don't really see the advantage to be honest. If I needed Asterisk access from anywhere I'd just run it locally on my laptop, connect to an ITSP via IAX or SIP, and run a softphone app locally. The only exception I can think of is when you'd want people to be able to leave voicemail on your
When I try this my GrandStream GXP-2000 gives me an error 603, which is
Declined.
-- Executing Pickup(SIP/8908-b7987738, SIP/[EMAIL PROTECTED]) in new
stack
== Spawn extension (internal, **8948, 1) exited non-zero on
'SIP/8908-b7987738'
-Original Message-
From: [EMAIL PROTECTED]
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