bilal ghayyad wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank or
company credit card provider?
How the user will
Philipp Kempgen wrote:
*snipped
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
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Philipp Kempgen wrote:
Richard Lyman schrieb:
Philipp Kempgen wrote:
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what ISNULL is for?
No. ISNULL() works on values
Godson Gera wrote:
On Mon, Nov 10, 2008 at 7:41 PM, David Budny [EMAIL PROTECTED]wrote:
What is the AMI command to see how many PRI channels are being used /
available?
Thanks
There is no direct command in AMI which will give you used channels number.
But you can easily keep
There are a few options.
He should probably start on the wiki.
http://www.voip-info.org/wiki/view/Predictive+dialer
Steve Totaro wrote:
If you can figure out how to generate .call files from your DB
entries, you have it made.
Vicidial needs alot of work as far as I am concerned, for free it
Octavio Ruiz wrote:
On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman [EMAIL PROTECTED] wrote:
Octavio Ruiz wrote:
On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan [EMAIL PROTECTED] wrote:
The output of a
CLI pri intese debug
at Asterisk CLI before make a test call would
Edwin Quijada wrote:
Hi!
I wanna know if here somebody has installed gnudialer ?
I installed but i dont know how to run it
Anybody has a cluee?
You would probably have more success reading all the README's and online
help.
If that does not provide the answer you can ask on the GnuDialer
michel freiha wrote:
Hi all,
I'm getting the following error when trying to make a PSTN call from
asterisk server:
end_sound = (null)
[Aug 19 20:51:17] WARNING[18945]: channel.c:3025 ast_request: No channel
type registered for ''
[Aug 19 20:51:17] WARNING[18945]:
Essien Ita Essien wrote:
Hi all,
I'm writing an application to Queue and Manage AMI Originate actions.
Basically, callfiles on steroids if you may :)
I'm facing the following challenges, and any ideas or pointers will be
hugely appreciated.
1. When I successfully Queue an Originate...
Ruddy Gbaguidi wrote:
Hi all
Back in the 1.2 days I think, there were some discussions about how two
asterisk
servers can share channel variables through an IAX protocol.
I don't see anything in 1.4 at least to be able to make it done.
Thanks
Back in 1.2 you had to use type 'friend' to
this is still true in up to asterisk 1.4.19
Ruddy Gbaguidi wrote:
It doesn't seems to be working ...
What I wanted to do is on the first server, Set a channel variable...
then dial the number.
When I received the call on the remote server, use that variable ...
Is it possible ?
Richard Lyman wrote
Doug Lytle wrote:
*snipped
exten = _X.,n,GotoIf($[${EXTEN:1} = 9]?not-parked,s,1)
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Nicholas Blasgen wrote:
Actually, I thought about it for a while. What I want is something
that will allow me to restart the message if another sound is
detected. Something like this:
exten = answermachine,1,Answer()
exten = answermachine,n,WaitForSilence(1000,2)
exten =
Veselin Kantsev wrote:
Thank you Tzafrir,
is there any way to make asterisk log the console input/output, itself?
I've only come accross methods of doing that using separate tools,
like tee.
look in the ~/.asterisk_history file
___
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better drop udp also.
Mark Hamilton wrote:
iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP
Good luck.
Via: SIP/2.0/UDP
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Matt Watson wrote:
On June 30, 2008 06:25:17 pm Mark G. Thomas wrote:
Hi,
After doing a yum update on my previously Centos-5.1 system, now
zaptel-1.4.11 fails to build with this below.
CC [M] /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included from
Mark Hamilton wrote:
Now you're just trying to get us all jealous, Steve. No good.
But I'd like that screwdriver!
I hope JT is taking notes and will get the higher ups to add 'tweakers'
to the digium store. G
On a personal note, i still haven't seen my 'sticker'! haha
TP'n to follow flow.
I haven't seen anyone mention ISNULL so there is an example (obviously
wrap it in the below Set(CALLERID(num)=...) as the other examples are)
${IF(${ISNULL(${CALLERID(num)})}?00:${CALLERID(num)})}
Eric ManxPower Wieling wrote:
This should do it, but I've not
David Ruggles wrote:
I'm trying to use the password entered with Authenticate to create dynamic
meetme conferences with the following dial plan:
exten = _XX18467,1,Authenticate(/etc/asterisk/meetme.pw|a)
exten = _XX18467,n,MeetMe(CDR(accountcode)) ; 281-8467
However
David Ruggles wrote:
I'm trying to use the password entered with Authenticate to create dynamic
meetme conferences with the following dial plan:
exten = _XX18467,1,Authenticate(/etc/asterisk/meetme.pw|a)
exten = _XX18467,n,MeetMe(CDR(accountcode)) ; 281-8467
However
Tilghman Lesher wrote:
*snipped
That is REALLY old, deprecated syntax, that will stop working in 1.6. The new
syntax is what everybody is already used to:
Yeah, well when it changes to and fro the only real constant is always
check the UPGRADE.txt for the version you use.
(as it will
blackwater dev wrote:
I'm head of RD for a dot com company and we are looking to create a
prototype using asterisk. Basically we people who visit our site and
search for goods listed by other people. Once something is found, a
phone number is listed in the results and person A calls
Vieri wrote:
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch = DUNDI/priv
exten = s,1,Set(CDR(userfield)=test)
exten =
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I
am
not able to get
Steve Shepherd wrote:
Greetings,
I've set up the AMI and am able to authenticate, however I am unable to
execute Action: Status. I get a permission denied error:
*snipped
read = system,call,log,verbose,command,agent,user
write = none
without the ability to 'write' a command, you
Olle E Johansson wrote:
*snipped
But on the other hand, if people rely on third-party distributions we
might want
to set up some kind of peer pressure on the maintainers - and possibly
identify them so we can support them and speed up their process.
/O
that is a very important, 'so we
Roger C. Beraldi Martins wrote:
Dear members of the list,
I have difficulties to obtain sync with a Digium TE420 PCI Express
For four entries E1, In my case I am with only 3 E1s available to
configure. The
telephony operator is BrasilTelecom and Signaling is R2 Digital.
*snipped
The
about how do this
configuration. I think it's something like this:
span=4,0,0,unused
that's it ? What do you think ?
If it's dosn't work I will check for de cable buildings.
Thank you for your help !
2007/12/3, Richard Lyman [EMAIL PROTECTED]:
Roger C. Beraldi Martins wrote
jumpers closed.
Thanks to all again, then I write with the result.
2007/12/3, Richard Lyman [EMAIL PROTECTED]:
#span=4,0,0,cas,hdb3
^ uncomment that
and
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
unused=94-124 #whatever your ending
Philipp Kempgen wrote:
Richard Lyman wrote:
I have never noticed, does the output of ztcfg change is it set to E1?
Yes. More channels. :)
Regards,
Philipp Kempgen
only if defined G
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TP'n to follow flow.
Seems lately (for me at least), if i did the pay on completion, i was
the one that got screwed over.
I obviously do not do that anymore. Sometimes you have to change your
methods regardless of your abilities.
Jon Pounder wrote:
Quoting Michelle Dupuis [EMAIL
Douglas Garstang wrote:
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
*snipped
Just for the record. I had to reconfigure the Sangoma startup scripts
even though my zap files were correct with d4,ami.
The real gotcha was in the cabling. One crossover was 1-5, 2-4, the
other 1-4, 2-5 and then straight through all required for the same
dialer. I have seen
Douglas Garstang wrote:
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output
applies to, to the start of each line? If you are proxying multiple systems,
how can it uniquely identify the output from each system?
Thanks,
Doug.
each Event block should have a
Bob Pierce wrote:
I've made a change to my manager.conf file in asterisk 1.2.18
Is there a way to reload that config file from the CLI without
restarting asterisk?
Bob
every time there is a new connection to the asterisk manager interface,
the manager.conf file is reread.
(meaning, it
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe EM.
I have done EM wink but have no idea about E4 Superframe EM and Google
is not helping me here.
Does anyone know about this type
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe EM.
I have done EM wink but have no idea about E4 Superframe EM and Google
Richard Lyman wrote:
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe EM.
I have done EM wink but have no idea
Brian Alexander wrote:
*snipped
The errors all seem to be about echo cancellation... What do I need to do to
force asterisk to never disable echo cancellation?
*snipped
there used to be this in ../zaptel/zconfig.h
#define NO_ECHOCAN_DISABLE
check if whatever version you are running has
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is
Jonas Arndt wrote:
Hi Guys,
I have already tried this one on the developers list. I have not been
successful getting much back there and I have notified them that I will
post this on the users list instead. Hopefully somebody have tried
something similar and can help out.
I am developing
a yellow alarm should mean there is a signal/encoding issue.
we used to see this on older gear when a single segment in the Rx leg
would flip from SF to ESF.
which means the old gear wanted SF, but all the newer telco gear
'defaults' to ESF nowdays, so there would be momentary yellow alarms as
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the addition of a
streaming music server (which, obviously, provides the MoH stream). Asterisk
should then simply pick up the stream play it whenever MoH is
Richard Lyman wrote:
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the addition of a
streaming music server (which, obviously, provides the MoH stream). Asterisk
should then simply pick up
Luki wrote:
Presently I have _all_ 900 calls blocked in Asterisk 1.25
but today I had to call a parts vendor at a 972 number.
Blocking anything with 9XX isn't a good idea. There are lots of
regular area codes in the 9XX block -- take a look:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to
Paul wrote:
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running
try using this in zaptel.conf
span=3,0,0,d4,ami
*snipped
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Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be in
CVS-TRUNK as an option for chan_zap
Eric ManxPower Wieling wrote:
Richard Lyman wrote:
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be
in CVS
Steve Edwards wrote:
*snipped
Psst -- don't tell the developers, but we could probably get something
similar to Asterisk with a box of tin cans, a spool of string and a
couple of carrier pigeons :)
don't forget the sneakers! G
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Tomislav Parcina wrote:
Richard Lyman wrote:
*snipped
fyi: manager originate is
channel + context + exten + priority
OR
channel + application + data
not both.
So, you are saying that this should look like this?
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Application: System
Data
Nathan Bell wrote:
I'm having trouble getting the manager interface to behave properly;
specifically the Originate event.
If I create an originate event as below, the calling phone will
auto-answer (as it's supposed to) but the receiving phone never rings.
It will timeout at 20 seconds.
Tomislav Parcina wrote:
Lee Jenkins wrote:
You have to login into the AMI server with proper credentials and
send commands.
*snipped
OK, maybe he doesn't show output, so I have tried this:
Action: Command
Command: ! rm /tmp/test.txt
Response: Follows
Privilege: Command
--END COMMAND--
But
Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote:
Hi Murphy,
Sorry! But I didn’t understand you :(
Can you give me an example?
When I talked about creating a new property in the events to return the
ActionID command, I just give an idea. My problem is to identify WHO has
raised
that there is no way to do it anymore.
Thanks,
[]’s
Moacir O. de Souza Junior
Belo Horizonte – Minas Gerais - Brasil
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Richard Lyman
Enviada em: quinta-feira, 29 de março de 2007 13:55
Para: Asterisk Users Mailing List
Alexander Lopez wrote:
It is a HUGE workaround but in concept it should work.
You will need to build completion confirmation into your script as you
will always get a success code from the manager.
Action: Originate
Application: System
Data: /path/to/script
Channel: Local/[EMAIL PROTECTED]
wrote:
*snipped
If I can't be confident enough in an important source of information like
this then I can't be confident enough to provide an Asterisk solution to
businesses. That's the way I see it. Yea, it's a wiki but it's the best
source of info out there.
*snipped
sorry to see you
Eric Bishop wrote:
show dialplan keeps showing contexts created by AEL. I tried
deleting /etc/asterisk/extensions.ael but kept getting these messages
in the Asterisk log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14
TP'n to follow flow
just like DNS, the 'root servers' would still see the high request hits,
prior to passing off to local caching app.
and *someone* must have this expense/headache to maintain them.
Natambu Obleton wrote:
Why not make it like DNS and have each provider have their lookups
Joe Greco wrote:
TP'n to follow flow
just like DNS, the 'root servers' would still see the high request hits,
prior to passing off to local caching app.
and *someone* must have this expense/headache to maintain them.
No, the root servers wouldn't. Please take a few moments to learn
Yuan LIU wrote:
On my wild learning curve, I encountered numerous occasions when a
channel remained in Congestion state after a Congestion() step
without going to the next step, which is Hangup(). I couldn't find a
definite pattern but it seems to happen when a channel is hung up by
the
Benny Amorsen wrote:
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL everytime you make a dns request, i agreed that it does not hit
RL the root servers, but every time you request a NON-cached one you
RL DO.
Nope. If you request foo.com and you have up to two days earlier
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL
TP'n to follow flow
or mod the /etc/asterisk/indications.conf
the /xxx is the duration (iirc)
example: busy is like 400/400,0/400
the /400 (each) is the duration
Eric ManxPower Wieling wrote:
Read the zapata.conf.sample file that comes with Asterisk
[EMAIL PROTECTED] ~]# grep toneduration
ManxPower Wieling wrote:
/etc/asterisk/indications.conf has nothing to do with the length of
DTMF tones sent out FXO ports.
Richard Lyman wrote:
TP'n to follow flow
or mod the /etc/asterisk/indications.conf
the /xxx is the duration (iirc)
example: busy is like 400/400,0/400
the /400 (each
Yuan LIU wrote:
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:58:08 +
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory
Pavel Jezek wrote:
I can confirm,
commands after Wait() are never executed in 'h' extension
and wait seconds argument in wait() is completely ignored
it's bug or feature? ;-)
h = {
NoOP(before ${EXTEN});
Wait(5);
NoOP(after ${EXTEN});
}
*snipped
in ael use WaitExten(5);
about executing diaplan when entering 'h'
extension, ie. after user hangs up phone...
and seems, something strange with processing wait() app in processiong
'h' extension in diaplan - timeout specified is ignored, and dialplan
stops processing
Richard Lyman wrote:
*snipped
in ael use
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set
*snipped
ast_set_variables(chan, vars);
insert pbx_builtin_var here --
ast_pbx_run(chan);
since DIALSTATUS and HANGUPCAUSE are both protected, you will probably
have to create another such as FAILEDCODE.
i hope
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and
Douglas Garstang wrote
*snipped
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
*snipped
a quick run through of of app_queue.c (my copy) for anything directly
dealing with a reload
shows tmp in use for realtime
later a reference for convert to
lenz wrote:
Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full
test of the viability of this solution before deployment, but I was
wondering if anyone has experience of such a setup and if there
*snipped
Second, when using a .call file (or the manager interface's Originate
action) the 'Dial' action is executed BEFORE entry into the dialplan, so
if it fails, nothing in your dialplan is executed and you get a somewhat
*snipped
not *exactly* true.
you need to add
;this extension
Benny Amorsen wrote:
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL grr, i hate when i typo (and reply to my own posts) exten =
RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx)
Heh, if you want to chase typos, perhaps you should add an underscore
before
Douglas Garstang wrote:
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies. When someone
calls a number in their own
Richard Lyman wrote:
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies.
When someone calls
Steve Murphy wrote:
*snipped
I've been fixing manager bugs here and there, and am willing to take on
any manager issues out there, for 1.4, and trunk, especially, so as to
have things nice and solid for 1.4 before it gets out of beta.
*snipped
Richard-- I'll lab up
1.4 and see if I can get
Douglas Garstang wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for
James Texter wrote:
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run sip show peer 2944093 in the
CLI, you'll see a blank line, followed by a line that is * Name. It
appears what you really want is a manager Action to
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Richard Lyman [EMAIL PROTECTED] wrote:
just wait till you get a 'hiccup' that causes a line to get cut off,
drop a char, and continue on next line. G
(examples below)
I've made heavy use of the Manager interface for over 2 years
-Useragent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098
Reg-Contact : sip:[EMAIL PROTECTED]
ChanVariable:
parkstart,10
Doug.
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Eric Bishop wrote:
You can't hanup channels with a call file you can only create them no?
*snipped
actually you could hangup a call using a call file
example
Channel: Tech/Dev-occurance
Application: Hangup
Data: somecausecodevar or digit equiv
Gregory Duchatelet wrote:
Hi all,
Another question for today, hope an answer for this one…
I have a program talking with asterisk via the AMI. I receive events,
and I would like to insert some events in the dialplan, which could be
catch by my program.
Any idea how to do this ?
Greg
Michael Collins wrote:
I’m interested in knowing if anyone else has worked around this issue:
I have an application that needs to check the status of the calls
going through Asterisk about every 5 seconds or so. I don’t want to do
“asterisk –rx ‘show channels verbose’” at the Linux command
Tzafrir Cohen wrote:
*snipped
Note that you better not use a terminal server settings. The SIP client
should run on the thin client's CPU, not on the server's CPU. The server
can help with the boot process (maybe a shared NFS root will prove
useful).
*snipped
that particular unit is also
Jay R. Ashworth wrote:
*snipped
The ability to detect precise SIT tones on placed calls would be
*really* good.
actually it is damn near impossible.
in a perfect world, if all the switch providers where adhering to ITU
spec on SIT's,
then it would be possible. they sad part is (at least
Tzafrir Cohen wrote:
On Tue, Sep 19, 2006 at 09:58:45PM -0700, mitcheloc wrote:
You are incorrect. The GUI you are referring to is the framework I already
mentioned. The webpages are static html javascript (AJAX functionality).
Asterisk has a simple built in HTTP server in trunk now which
Jan Fousek wrote:
Hi all,
is there any possibility of setting the global variables from outside of
asterisk?
Like manager api or something like that.
Thanks a lot
not sure about current svn trunk,
but in the past you could set a channel var with
action: SetVar
channel: Zap/49-1
MF wrote:
Has anyone got a clue about this?I need to know which operator to
send a message to, prior to the queue command ringing him, (just
after he is assigned)
Anyone knows if I can get to know the operator ACD choosed to send
the call by using Realtime Queue, or maybe via the
Jeremy McNamara wrote:
Douglas Garstang wrote:
Oh, and I see nufone caters to residential. We only cater to business
customers, who's needs are a lot more demanding.
Apparently you haven't actually gone to our website which, since you
brought it up, will be re-launched on September 5th, 2006
Stefan Reuter wrote:
Johannes Zweng wrote:
Although I can associate every incoming event to a specific channel on
Asterisk (because of the Uniqueid field) I see no possibility to identify
without doubts which channels were created as a result of my Originate
action.
add an ActionId
Tony Mountifield wrote:
*snipped
Comparing with 1.2, I see there were originally two calls to manager_event(),
one for OriginateFailure and another for OriginateSuccess.
They have now been combined into one, with a conditional event name,
which may have given rise to the mistaken impression
*snipped
Patrick, yes, this is a literal portion. I have no reason to believe that
spsaces between the priority, and the command cause problems, so I haven't
tried that yet. Just trying to make the horrible assembler-like Asterisk
dialplan language more readable.
*snipped
this doesn't
Bill Schaffer wrote:
I don't know the Cicso equipment, but an educated guess from the
configuration info you gave tells me that you need to use MFC R2
signalling.
Read about it here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2
And pay heed to the big disclaimer at the front
# This script asks asterisk to rotate its logs on its own.
postrotate
/usr/sbin/asterisk -rx logger rotate
endscript
is what we use and it seems to be just fine.
(logger reload reopens the log files, where logger rotate,
rotates then then reopens)
Matt Florell wrote:
Welcome
*snipped
I like the idea that I can just pick up the phone, dial an extension, record
an announcement, and be sure that announcement will play during extended
hold times.
Thanks for the ideas, Richard!
It just came to me, about using timeouts on the queues, play an announcment,
update the
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