It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan.
snip
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
It's the quotes that are messing it up... what you probably want
You could also use the cid syntax in the extension
exten = s/ObnoxiousCallerId,1,Goto(getlost)
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is
Thanks for the help on this thread all.
It would make sense if I write an AGI and incorporate a DB backend to
check against numbers I want explicitly dropped. If anyone has such a
utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
it up and probably provide a web frontend for
Dear all.
I have what appears to be a configuration error but I cannot for the life of me
see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
Unable to make outgoing calls via E1 euroISDN Digium
On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote:
Dear all.
I have what appears to be a configuration error but I cannot for the life of
me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
PROTECTED] On Behalf Of Matt Scott
Sent: 22 May 2007 13:11
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial out issues.
Dear all.
I have what appears to be a configuration error but I cannot for the
life of me see what it is. (I am a newbie)
I have searched the wikki
On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
Good catch Morgan!
Hi,
I am in the process of planning a dial plan, In regards to the
requirement, I am confused how to go about the dial plan.
The scenario is like below.
BRANCH - A - (COMPANY)
Line 1 -- Extension 239
Line 2 -- Extension 8239
BRANCH - B - (COMPANY)
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
--
Thanks
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asterisk-users mailing list
To UNSUBSCRIBE or
On Thu, 19 Apr 2007, [EMAIL PROTECTED] said something to this effect:
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
Use IF conditionals in the dial plan to manipulate strings and/or create
new dial
[EMAIL PROTECTED] wrote:
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
Here are a couple of good reads for you
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Planning
Hi there,
I'm converting a dialplan callback type application to fastagi as I'm
hitting the buffers with respects to getting useful results from CDRs.
It works by a spool call file triggering a Local extension, that extension
then does the first dial to a client. I dial to a local context
Hi,
While Dial rings can a caller press 0 (or other number) to leave a
voicemail? I found that with a # can transfer to different context. I
want to use that two features together.
--
Suich
___
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Hi All,
Customer is requesting 1 incoming toll free #, that dial out to 4
different terminating numbers, not ring all at once but ring #1, then
#2, then #3, then #4, then back to #1 consecutively on inbound calls,
regardless if someone is on #1. So this is not like a hunt group,
more like
Hi All,
Customer is requesting 1 incoming toll free #, that dial out to 4
different terminating numbers, not ring all at once but ring #1, then
#2, then #3, then #4, then back to #1 consecutively on inbound calls,
regardless if someone is on #1. So this is not like a hunt group,
more like an
Hello,
I've seen this already asked and answered but it is still a no go for
me.
I'm trying to do some preprocessing in the middle of a call, before
bridging.
I've seen two choices: M() and G() parameters of the Dial() command.
G() was discarded because I don't know if it is possible to
OK, but y would i want to use it. i mean y not use goto and y this? and what
dialout files are you talking about?
On 3/20/07, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
This is a tool that allows you at any time and any place of your
Dialplan or Dialout Call file to dial a specific extension at
HI,
I dont understand the syntax of the dial application when used like this:
Dial(Local/[EMAIL PROTECTED])
i want to know what is this Local doing instead of Tech like SIP, IAX,
H323?
--
Regards
Rizwan Hisham
Software Engineer
___
--Bandwidth and
Rizwan Hisham wrote:
I dont understand the syntax of the dial application when used like this:
Dial(Local/[EMAIL PROTECTED])
i want to know what is this Local doing instead of Tech like SIP, IAX,
H323?
SIP/200 would dial a device (the SIP user 200) whereas
Local/200 dials the extension
Hi,
This is a tool that allows you at any time and any place of your Dialplan
or Dialout Call file to dial a specific extension at a specific context,
even if you are not currently in the specific context.
example:
you are at [from-internal] context and you can say:
[from-internal]
exten=
you should separate to two lines, like...
exten = _366[5-9]X,...
exten = 36700,...
Hall, Eric M. wrote:
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 - 36700 to a Context 'test' however I'm only
able to get 10 to work at a time. Any ideas?
Any help would be great!
Hall, Eric M. wrote:
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context ‘test’ however I’m only
able to get 10 to work at a time. Any ideas?
The square brackets
The second example is for a four digit extension. while the first is
for a five digit extension. everything within the brackets is meant
for just one digit.
On 3/3/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this
This worked:
Dial(SIP/[EMAIL PROTECTED],,D(12345678))
however, the problem now exists in the disconnection. Asterisk tries to
bridge the call, play dtmf but never disconnects. What is there a specific
syntax to the D command that specifies a disconnect period.
I am thinking a better solution
this dials, and upon answers plays dtmf tones, but does not auto disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,S(8),D(12345678))
any ideas of what wrong
Supa wrote:
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,S(8),D(12345678))
any ideas of
Thanks that worked, but it still tries to bridge call after dtmf, then fails
instead of moving on to next number to dial and page
On 2/25/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Supa wrote:
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten =
Supa wrote:
Thanks that worked, but it still tries to bridge call after dtmf, then
fails instead of moving on to next number to dial and page
So tack on a g to the end of your dial strong, to continue along the
dial plan upon disconnect.
___
From: Supa [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 15:45:08 -0500
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten =
23 feb 2007 kl. 14.06 skrev ast guy:
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
Yes, Asterisk is a multiprotocol Open Source PBX. Those
functions
Probably just a simple syntax issue, but does anyone know how to dial a
number and the once phone has been answered, play DTMF tones and then
disconnect. I am trying to use this for page notification.
Ive been trying the following string with out luck:
exten =
/5198881212www12345678)
also, you were missing a right parenth.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Supa
Sent: Sat 2/24/2007 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] dial a pager and enter DTMF
Probably just a simple
Discussion
Subject: [asterisk-users] dial a pager and enter DTMF
Probably just a simple syntax issue, but does anyone know how to dial a
number and the once phone has been answered, play DTMF tones and then
disconnect. I am trying to use this for page notification.
Ive been trying the following
is there a way to pipe the dial command with SendDTMF(123456)
What I am trying to do is dial an extension and have it page a group of
pagers with the same number. Saving a lot of time over dial each one
manually by hand.
___
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Buy a cap code from the paging provider and program that cap into the
group of pagers that way when you page that cap code all of the pagers
will trip.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Supa wrote:
is there a way to pipe the dial command
Supa wrote:
Probably just a simple syntax issue, but does anyone know how to dial
a number and the once phone has been answered, play DTMF tones and
then disconnect. I am trying to use this for page notification.
Ive been trying the following string with out luck:
exten =
From: Supa [EMAIL PROTECTED]
Date: Sat, 24 Feb 2007 10:05:06 -0500
Probably just a simple syntax issue, but does anyone know how to dial a
number and the once phone has been answered, play DTMF tones and then
disconnect. I am trying to use this for page notification.
Ive been trying the
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
-ag
___
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On 18:06, Fri 23 Feb 07, ast guy wrote:
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
yes. It's a function in asterisk call thingie, not in the
sip
Hi Roy,
If its perl script,you can try this.
use Asterisk::AGI;
our $AGI = new Asterisk::AGI;
$AGI-EXEC('Dial', 'Zap/g2/8005551212');
On 2/11/07, Roy Kidder [EMAIL PROTECTED] wrote:
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part I can't seem to get right is the dial function. I tried to
mimic the dial plan
From: Roy Kidder [EMAIL PROTECTED]
Date: Sat, 10 Feb 2007 23:15:07 -0500 (EST)
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part
Hi Roy,
Look I dont know why u specify 'zap/1-1', but i do things like this on
my agi scripts a lot of times:
...
$stdin= fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/tmp/outPUT.log', 'a');
...
fwrite($stdout,EXEC DIAL
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi people.
I'm hoping someone has come across this problem with version 1.2.14
In my dial plan I call various SIP phones using the following little
macro:
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers
Can I ask for some advice on dial-plan construction please
I have setup my dialplan to use 9 to get a zap trunk, leaving everything
else for internal extensions.
However, this creates a problem in that my callerid is correct, but
doesn't work to re-dial the incoming caller. So if I simply
I don't know about SNOM, but with Xlite Softphone you can have the SoftPhone
internal dialplan.
Ex.
[29];match=1;pre=0; this adds a Zero to every nine digits number
s I dial begining with 2 or 9 , this has nothing to do with asterisk, is
VoiP phone dialplan.
So you can tell to the
On Tue, 23 Jan 2007, Ed W wrote:
Can I ask for some advice on dial-plan construction please
I have setup my dialplan to use 9 to get a zap trunk, leaving everything else
for internal extensions.
However, this creates a problem in that my callerid is correct, but doesn't
work to re-dial the
Hi
There was a thread about this not too long ago, so the archives may
have a bit more on it...
The way I handle it is by forcing the caller to dial the full number
starting with zero (normally 10 or 11 digits in the UK - which I'm
guessing you're from too)
Yes, I use something similar
Dial(...|30|g) does not seem to work
whereas
Dial(...|30|gh) works just fine
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
--Bandwidth and
Hi all,
Can someone point me in the right direction here. What I'd like to do
with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom
phones and after the 3rd digit is entered, it dials that extension and
b) dial 9 to get out like older PBX systems. Since my internal
extensions
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to support an extra digit when
dialing 9.
On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote:
Hi all,
Can someone point me in the right direction here. What I'd like to do
with
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an example of this on the web but I can't seem to find it.
Any advice
: [asterisk-users] Dial own extension to get to voicemail.
I've gotten this Polycom 501 pretty much licked, but I need to know if there's
a way in a dialplan to say if someone dials their own extension it goes
straight to voicemail and asks them for their password. I thought I saw an
example
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an
Is this what you are looking for
exten = _9.,1,Set(CALLERID(num)=3045551212)
exten = _9.,n,Dial(ZAP/g2/${EXTEN:1})
On 12/20/06, Bruce Reeves [EMAIL PROTECTED] wrote:
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to support
[default]
Some extensions defined
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I have the above defined in extensions.conf. This enables me to make
outgoing
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten =
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Oups, pressed Send too fast, here is take 2
exten = _90[1-9].,1,Dial(IAX2/[EMAIL
Am Sonntag, den 17.12.2006, 18:11 -0500 schrieb Time Bandit:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten =
I have 4 Polycom phones with multiple line keys so multiple incoming
calls work fine
The way I would like the incoming call flow to work is as follows:
1) 2 groups consisting of 2 phones each
2) Incoming call rings the first group, if no answer, the 2nd
group is rung
3)
Dial() cmd seams unable to detect caller hangup?
so if the call file land in a exten, for example:
[callfile-landing]
exten=1,1,dial(SIP/XXX)
exten=1,n,hangup
when caller after conversation and hangup, the dial cmd is unable to detect
that and it will ring the caller and called party 2
Try the background() command to play a sound file or say digits to the
user, while not holding up the dial-plan.
On 11/13/06, Matthew Rubenstein [EMAIL PROTECTED] wrote:
I initiate a call with a callfile, specifying the From phone# as the
channel Dial(), and the To phone# as the
Hi,
I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a
user via SIP and playing a reminder file when the user picks the phone.
I use Gizmo service for SIP and I'm able to call through it.
However, when asterisk dials a number, Gizmo first answers then tries
Put canreinvite=no in asterisk sip user extension.Someprovidersdonotsupportreinvitesandhenceyougetsilenceiguess. On 13/11/06, Yuri Veremeyenko
[EMAIL PROTECTED] wrote:
Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file
I initiate a call with a callfile, specifying the From phone# as the
channel Dial(), and the To phone# as the Extension Dial(). I announce
the To phone# to the From listener with the A() option to the Dial()
command. It seems that the A() app plays audio while blocking return
from the From
Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten
Thanks, that set off a light bulb In my spa3K my incoming dialplan was
set to (S0:405)
Since this is a one FXO unit and my [from-pstn] will always be that line
can I make it generic and use the 's' extension as I described? If so what
would that spa3k dialplan be? just s0 ?
Doug
On Tue, 7
Answering my own question. If you want to connect an spa3K with
generic pstn inbound do the following...
for the pstn to voip dialplan in the pstn tab - (S0:ip-address-of-*)
in sip.conf
[sipurafxo]
context=from-pstn
etc.
Then in * extensions.conf use the s extension.
[from-pstn]
- Original Message -
From: BerkHolz, Steven [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 01, 2006 1:27 AM
Subject: [asterisk-users] dial D option with w for wait?
From WIKI:
D(digits): After the called party answers, send digits as a DTMF stream
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten s,2,playback(blah)
etc.
It never answers but if I do this
[from-pstn]
exten
From WIKI:
D(digits): After the called party answers, send digits as a DTMF stream,
then connect the call to the originating channel. (You can also use 'w'
to produce .5 second pauses.)
When I use the D option to send a call to my paging system and pick a
zone, the Tone is too early.
I have
There was patch for 1.0.x version of Asterisk that is quite useful. Is there
patch for 1.2.x version and will this i parameter be in 1.4.x version of
Asterisk?
Have a nice day!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP:
Hello.
I have a TDM 2400P and havent figured out how to
attach a phone to one of the FXS channels in the bank and dial out. To dial in
the analog phone is easy, all I had to do was to insert a line in the
extensions.conf saying exten = 430,1,Dial(Zap/17,20,t). But I cant
figure out how
Hi
Is it possible to have Asterisk dial an external number without having a phone?
I want to make a box that can generate calls into a normal PABX and
just play MOH or similar.
It's for stresstesting applications I'm developing.
If it could be done via the Manager interface, it would be
This is strange. I upgraded from an older [EMAIL PROTECTED] that was
working to the latest Tribox. I also added a A204 board, but for some
reason neither the Grandstream phone or a phone connected to the Linksys
ATA has any audio either way via the Telasip connection. Audio works OK
between
On 1 Oct 2006, at 04:54, Naija Man wrote:As a habit, I do not force users to dial 9 or any other prefix of any kind to access external lines. You can just check the dialled number and prefix with appropriate digits appropriately. See below. NOTE: THIS IS US-CENTRIC!! but can be easily made to work
On Sun, 1 Oct 2006, Matthew Thompson wrote:
In the UK we have 6,7,8, and 11 digit local dialling and 8, 10 and 11
digit national dialling.
You forgot 5 digit local numbering ;-)
A town near me has 5-digit local numbers, my town has 6 digit local
numbers and we both have the same 5-digit STD
Naija Man wrote:
As a habit, I do not force users to dial 9 or any other prefix of any
kind
to access external lines. You can just check the dialled number and prefix
with appropriate digits appropriately. See below. NOTE: THIS IS
US-CENTRIC!!
but can be easily made to work for any country.
On Sun, Oct 01, 2006 at 09:42:32AM -0500, Eric ManxPower Wieling wrote:
My problem with this type of dialplan is that users must wait for
DigitTimeout before the call is processed.
I was going to ask about that.
What's the common value for that number, and secondarily, does Asterisk
support
Jay R. Ashworth wrote:
On Sun, Oct 01, 2006 at 09:42:32AM -0500, Eric ManxPower Wieling wrote:
My problem with this type of dialplan is that users must wait for
DigitTimeout before the call is processed.
I was going to ask about that.
What's the common value for that number, and secondarily,
On Sun, Oct 01, 2006 at 03:54:49PM -0500, Eric ManxPower Wieling wrote:
Jay R. Ashworth wrote:
What's the common value for that number, and secondarily, does Asterisk
support the traditional #-cutthrough to signify that you're done
dialling, as LEC switches generally always have?
You can
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote:
Here in the UK, I've installed several small systems without a dial-9 for
an outside line type thing. The outside line prefix is effectively digit
zero. (which is preserved and dialled on the outgoing zap lines)
There is an
On 30 Sep 2006, at 19:35, Jay R. Ashworth wrote:
I know that this has been a problem for traditional PBXen for years,
and the only solution I've ever been able to see is use 8 as your
outdial prefix... but no one seems to ever do that, even 20 years on.
Never say no one. Our legacy PBX is
On Sat, Sep 30, 2006 at 02:35:49PM -0400, Jay R. Ashworth wrote:
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote:
Here in the UK, I've installed several small systems without a dial-9 for
an outside line type thing. The outside line prefix is effectively digit
zero. (which
On Sat, Sep 30, 2006 at 10:39:09PM +0300, Tzafrir Cohen wrote:
I know that this has been a problem for traditional PBXen for years,
and the only solution I've ever been able to see is use 8 as your
outdial prefix... but no one seems to ever do that, even 20 years on.
Is this really not
-- Forwarded message --From:Jay R. Ashworth
[EMAIL PROTECTED]To:asterisk-users@lists.digium.comDate:Sat, 30 Sep 2006 14:35:49 -0400
Subject:[asterisk-users] Dial-9 (was Extension Numbering)
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote: Here in the UK, I've
David Gagnon schrieb:
Are you having this problem with an analog line or PRI ?
David
Sorry, forgot to include that information: It's a PRI.
My Asterisk is: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q with Zaptel 1.2.6.
Tobias
___
--Bandwidth and
Hi,
we have experienced som troubles with the timeout option of the
Dial-App. It seems the Dial startts counting down the timeout imediatly,
but there are great differences when the called phone actually starts
ringing. If i call a landline phone in my own country it is nearly the
same, but if i
Are you having this problem with an analog line or PRI ?
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tobias Wolf
Envoyé : 18 septembre 2006 11:41
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Dial
Hi list,
I've got following Problem:
i have severel phones on my asterisk. and externel lines connected (POTS
sip, does not matter)
a externel caller A (CID(num)=0815) calls me ( 4711) .
4711 can be distributet to severel internal extensions for example 23
and 42.
23 is on ZAP/1 and 42
We use our own CDR, but as I understand, the C option resets the CDR,
that does not means is not going to save cdr, but is going to restart
the CDR. So, a simple NoCDR() before dialing should work, or ForkCDR()
and then NoCDR() if you want to save previous data.
Regards
On 8/27/06, Master Abi
When I use
exten = _70XX,1,NoCDR()
exten = _70XX,2,Dial(SIP/${EXTEN}|20|tr)
I get
Executing NoCDR(SIP/7002-081ac898, ) in new stack
Aug 28 15:27:18 WARNING[4670]: cdr.c:443 ast_cdr_free: CDR on channel
'SIP/7002-081ac898' not posted
Aug 28 15:27:18 WARNING[4670]: cdr.c:445 ast_cdr_free: CDR
Normal behaviour since the call record before executing NoCDR() was
not posted (saved)
Regards
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
When I use
exten = _70XX,1,NoCDR()
exten = _70XX,2,Dial(SIP/${EXTEN}|20|tr)
I get
Executing NoCDR(SIP/7002-081ac898, ) in new stack
Aug 28 15:27:18
That is what I thought, but then how do I STOP recording CDR's. If I use
it in the h extension, it also gives a warning.
Moises Silva wrote:
Normal behaviour since the call record before executing NoCDR() was
not posted (saved)
Regards
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
When I
just ignore the warning, no CDR will be saved
On 8/28/06, Master Abi [EMAIL PROTECTED] wrote:
That is what I thought, but then how do I STOP recording CDR's. If I use
it in the h extension, it also gives a warning.
Moises Silva wrote:
Normal behaviour since the call record before executing
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
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