Sure, just make your voicemail wait 5 seconds before
answering the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Navneet
ShahSent: Thursday, March 16, 2006 10:45 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dialplan :
Forwarding call to voicemail
Subject: Re: [Asterisk-Users] Dialplan woes
Hi Gabe,
The issue was because I didn't load pbx_config.so in modules.conf :)
Thanks,
Dave.
On Sun, 2006-03-12 at 13:15 -0800, Gabriel Afana wrote:
Hi,
After updating your sip.conf and extensions.conf, did you reload
asterisk
Hello all,
Inspired by the Asterisk talks at FOSDEM 2006, I've decided to give it a
whirl. I'm having some newbie problems with my dialplan and was
wondering if anyone could be of assistance Smile
When trying to dial 500, 600 or 601 I get the following notice:
pbx.c:1330
Hope [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, March 12, 2006 2:31 AM
Subject: [Asterisk-Users] Dialplan woes
Hello all,
Inspired by the Asterisk talks at FOSDEM 2006, I've decided to give it a
whirl. I'm having some newbie problems with my dialplan and was
wondering
After updating your sip.conf and extensions.conf, did you reload
asterisk? Asterisk caches the config files and does not re-read them unless
you issue a sip reload, extensions reload or an all-in-one restart when
convenient at the CLI.
Actually, the all-in-one is done with only reload, no
:37 PM
Subject: Re: [Asterisk-Users] Dialplan woes
After updating your sip.conf and extensions.conf, did you reload
asterisk? Asterisk caches the config files and does not re-read them
unless
you issue a sip reload, extensions reload or an all-in-one restart
when
convenient at the CLI
unless
you issue a sip reload, extensions reload or an all-in-one restart when
convenient at the CLI.
- Gabe
- Original Message -
From: Dave Hope [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, March 12, 2006 2:31 AM
Subject: [Asterisk-Users] Dialplan woes
SellVoIP appears to follow a US dialplan. A US numberis
dialled as 1NXXNXX whereas an international (to the US) numberis
dialled as 011X.
Frankly, I didn't ask whether international numbers like
Barbados where the code remains as 1 butare international (to the US) need
the 011 or can be
At 07:57 AM 03/05/2006, you wrote:
How can I strip the 00 and insert 011 in one entry in the
dialplan. I'm stripping the 00 and passing the rest of the numbers
for numbers dialled as 001X. (as in: 00|1XX.) but in case of
numbers out of the US, how would I insert the 011 ?
exten = _011X. ,
(been googling all day) - so much
appreciated.
- Original Message -
From: Ira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 05, 2006 21:01
Subject: Re: [Asterisk-Users] Dialplan - strip IDD prefix
Hello,
I'm interesting with asterisk, my plan is replacing our PBX office with
asterisk, now i've AAH and it's worked. Now i have a question, how can
limit the user to call international calling, linterlocal calling, and
mobile phone calling.
international calling started with = 00 in my
I need to add the facility to allow some of my extensions to be able to
dial toll calls by entering a Pin Number to enable toll calling. For
example dial
*331234567 from any extension to enable Toll calling from extension
123(pin 4567)
*34123 from any extension to toll bar extension 123
would
On 11/25/05, Daniel Wright [EMAIL PROTECTED] wrote:
Steve Davies wrote:
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten = _X.,1,NoOp()
Would trigger for either a
hi,
can anyone please guide me as to how i can implement this in extensions.conf:
my PSTN line normally has its longdistance capability locked which can be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long distance using the
zap channel, how can i
hi,
can anyone please guide me as to how i can implement this in
extensions.conf:
my PSTN line normally has its longdistance capability locked which
can
be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long distance using the
zap
channel, how
thanks steve,
the reason i cannot remove the restriction on the telco line is that an
analog fone is connected to the phone jack of the x101p and some
visitors occasionally use the fone and they're supposed to only call
local toll free numbers.
Your suggestion of doing the restrictions within
How about this?
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
-or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
I have not seen restrictions set before dialing since local numbers
would not fall under the restriction but that is what you said. Usually
you dial the number
Thanks Steve,
But this will not work for me because after yourcodehere the line
will give a confirmation tone (similar to a congestion tone only
faster) then after flashing or certain period will turn into a busytone
and to get the dialtone again i need to Flash again before i can dial
${EXTEN}.
Wow, what a pain. I would just pickup an FXS and be done with it.
Thanks Steve,
But this will not work for me because after yourcodehere the line
will
give a confirmation tone (similar to a congestion tone only faster)
then
after flashing or certain period will turn into a busytone
This might work if you switch it around a little.
http://www.voip-info.org/wiki-Asterisk+cmd+Flash
-Original Message-
From: MZ [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 27, 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Yeah, and unlocked ATAs are not available in the market here.
I even had to pay almost twice the cost of my X101p clone for shippingOn 11/28/05, Steve Totaro [EMAIL PROTECTED]
wrote:Wow, what a pain.I would just pickup an FXS and be done with it.
Thanks Steve, But this will not work for me
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten = _X.,1,NoOp()
Would trigger for either a single digit, or for a longer number (as
long as it starts with a digit)
In practice
Steve Davies wrote:
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten = _X.,1,NoOp()
Would trigger for either a single digit, or for a longer number (as
long as it starts with a
On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote:
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
Has anyone built a game with the dialplan? I would think this would
most easily be managed by an AGI, but its possible with realtime
extensions.
On Sat, 2005-09-24 at 23:30 -0700, trixter http://www.0xdecafbad.com
wrote:
On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote:
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
Has anyone built a game with the dialplan? I would think this would
Has anyone built a game with the dialplan? I would think this would
most easily be managed by an AGI, but its possible with realtime
extensions.
The game would be like 'adventure' that I first played on a prime in
1979. Or any of the infocom games (ie zork). Infact since the infocom
spec is
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
Has anyone built a game with the dialplan? I would think this would
most easily be managed by an AGI, but its possible with realtime
extensions.
The game would be like 'adventure' that I first played on a
I am trying to figure out how to try different VOIP providers if they
aren't able to terminate the call because they don't offer service to
that dialing area.
The error that gets logged to the console is:
Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call
rejected by x.x.x.x: No
i guess is usefull a neighcompany context, where you will allow users
to call other companies, using a company prefix. I need more info about
your real dial patterns in order to suggest something more specific.
best regards
On 9/13/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have to design
Would like to use # and * to add and remove queue members like this:
exten = #14,1,AddQueueMember(queue_test1)
exten = #14,2,PlayBack(agent-loginok);
exten = #14,3,Hangup
exten = *14,1,RemoveQueueMember(queue_test1)
exten = *14,2,PlayBack(agent-loggedoff)
exten = *14,3,Hangup
The problem is I
I have to design a dialplan for mulitple contexts (multiple companies)
and I'm not sure how to go about it and I thought someone may offer
help. Here is some background. There are three separate companies,
let's say A, B and C. Each has their own context and each has their own
set of numbers
The IP - pbx extension calls are already workin fine.
Now Im just configuring the pbx extension - IP calls this way:
[pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip
clients]
Thats why the Dial is for SIP only.
Now Im going to try to get the 118 in Asterisk, because the
I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt
working. Now, to implement my dialplan in witch all the SIP phones are
74XXX, I must put the 74 manually, and the line is:
exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)
Thank you to everyone that helped me.
Joao Pereira wrote:
I got it
The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt
working. Now, to implement my dialplan in witch all the SIP phones are
74XXX, I must put the 74 manually, and the line is:
exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r)
Don't use r.
r:
Ok, but thats static routing. My architecture is this:
[pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip
clients]
I can't put in Asterisks sip.conf the hundreds of pbx extensions (and
they are always changing), I must do a dinamic forward for all 74XXX calls.
I think
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Just an observation that you have an invalid address there; you have
1193 instead of 193 I believe. Fix this and I see no reason for your
problem to remain.
--
-Bryce
[EMAIL PROTECTED]
But to have a transparent integration with VoIP and legacy, I cant make
users dial twice... or having to whait for Asterisks dialtone, and dial
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and
the IP phone rings.
Asterisk just need to forward the 74XXX calls,
yes, I know, in my extensions.conf is writen correctly.
Thanks
Joao
Bryce Chidester wrote:
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Just an observation that you have an invalid address there; you have
1193 instead of
.).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Wednesday, August 10, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dialplan defenition
But to have
Ok, I m getting to the point,
This route:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Isn't working because the dialed number isnt maching _74XXX
I putted Asterisk in capi debug mode and when I dial 74118 he says:
gnugk*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND ID=001
On Wed, 10 Aug 2005, Joao Pereira wrote:
Ok, I m getting to the point,
This route:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Isn't working because the dialed number isnt maching _74XXX
I putted Asterisk in capi debug mode and when I dial 74118 he says:
gnugk*CLI capi debug
Joao,
I don't think that number 81 is part of the dialed digits. Maybe this is an
ID of this or something like this.
I think that asterisk is not recognizing the first 2 digits, and passing just
the others maybe is something related about ignorepat (like a don't
ignore pattern ?).
Hello All,
Right now I have several providers. Voipjet, Teliax, and more recently
Broadvoice.
Broadvoice gives me unlimited to europe, but what I want to do is
determine the best way to setup a dialplan so for example, certain
countries will go through the cheapest route.
I am really only
Good day,
I would recommend using an LCR engine to do this. There is at least one
listed in the wiki. I am also nearing completion of an lcr engine that
integrates with ASTPP, asterisk billing software. It will be easy to
setup once I get it working. :-)
Darren Wiebe
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
This is a similar idea to LCR (least cost routing) on normal pbx
systems.
Any advice would be nice, since I'm sure those users who use asterisk
for more commercial purposes have figured our a way to do this...
Jump to the LCR section on this page:
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that
Most of the documentation I have read through shows dial plan examples that
dial the SIP phones and stop if one is picked up. I have not seen an example
of or read how to stop the SIP dial when an analog phone is answered.
How can the extension be set up so that when an analog phone is picked up
the problem is how are you getting there? i mean, what do you have in
sip.conf and please post all the relevant text in extensions.conf, not
just the 'exten = blah' part, we need to know context names to see if
its matching the sip.conf configuration
regards
On 7/28/05, Joao Pereira [EMAIL
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
but this way all calls go to [EMAIL PROTECTED]
Joao Pereira schrieb:
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
but this way all calls go to [EMAIL
I had tried that also, but it didnt work. In that case, if I dial 74118
(for example) Asterisk answers this:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid
extension 's' in context 'default', but no invalid handler
I think it needs the s... but how do I put the s and
the problem is that you are using the 's' extension. If you want to
match, as you said, the numbers like 74XX, then you should put
something like this:
[sipextens]
exten = _74XX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
in this way, all the numbers starting with 74 followed by 2 more
numbers, will be
So just don't send them to extension s but extension _74XXX
Christian
Joao Pereira schrieb:
I had tried that also, but it didnt work. In that case, if I dial 74118
(for example) Asterisk answers this:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid
extension 's' in
On Thu, 28 Jul 2005 10:30:15 +0100
Joao Pereira [EMAIL PROTECTED] wrote:
snip
Then I tried:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
I like to do this:
** extensions.conf **
[globals]
MYSIP=SIP/mysipphone
[mycontext]
exten = _74XXX,1,Dial(${MYSIP}/${EXTEN})
;exten =
In the dial plan, with GotoIf...is there a way to do a logical
negation(NOT)? I want to check if the year is a leap year and the below
code is how I would write the check in C. According to the wiki there
isn't a logical NOT as a valid expression. From reading the
I wanted to use AgentMonitorOutgoing(c) to know which agent made an
outbound call. Its supposed to record the agent id to the channels
column in the CDR, but it doesn't.
I put it on priority one, after an agent makes an outbound call
Does anyone have an example that uses this feature and works?
Hello!
Following the instructions on voip-ip.org I have implemented Realtime
with MySQL for my Asterisk server. The individual extension
configuration is managed in a table called extensions.
Still I have to keep some data in the extensions.conf, namely the switch
and the include statements. Is
On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote:
Hello!
Following the instructions on voip-ip.org I have implemented Realtime
with MySQL for my Asterisk server. The individual extension
configuration is managed in a table called extensions.
Still I have to keep some data in the
Sometimes for me unknown reasons a wakeup call cannot delivered to a
phone and ends up in the voice mail box (and consequently sent via email
to the phone user).
It would be nice to find the reason why the phone was not reachable, but
for sure it is useless to send a wakeup call to the
Title: Normal
If someone has a minute, I would appreciate their help
configuring my dialplan. I am using 2 Sipura-2000s to connect to the CO
ports on my legacy PBX. Im tyring to setup the dialplan so that
when someone enters an extension (1XX), it will determine which of the 4 sip
Hello,
I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI
as channels. A call comes in via IAX2 and should be redirected to CAPI.
So I wrote the following dialplan:
[fromiax]
exten = _8XXX,1,Answer
exten = _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r)
[fromcapi]
exten =
You could try the new chan_capi-cm-0.5.1 in which you dont need to have any
msn defined in capi.conf
On Wed, 22 Jun 2005 12:28:58 +0200 (MEST), Patrik Schindler wrote
Hello,
I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX
and CAPI as channels. A call comes in via IAX2 and
On Wed, 22 Jun 2005, jurczak wrote:
You could try the new chan_capi-cm-0.5.1 in which you dont need to have any
msn defined in capi.conf
Thank you very much, this solved my problem!
Do you know a solution for non-capi but i4l devices? It's not an error but
a warning only, so it's not a real
In realtime extensions the pipe is the separator. And I see a number
of commands that also use the pipe as a separator in regular
extensions.conf.
Can the pipe be used universally instead of a comma as a separator?
Chris
___
Asterisk-Users mailing
Starting with Asterisk makes fun, ... and soon you have a huge dialplan.
How are you going to organize it so that it is easy to maintain? It
comes even more complicated when you put the dialplan into Realtime, ...
and that is my goal now, to prepare the dialplan for realtime optimized
and for
I recently
updated my sip.conf and extensions.conf files and after
shutting
down asterisk and restarting it (asterisk -cvvv)
it shows and
empty dialplan (show dialplan)
*CLI
show dialplan-= 0 extensions (0 priorities) in 0 contexts.
=-
What could
cause somthing like this
below is a
What I want is for an incoming call to ring for say 20 seconds, then
hangup, then call an external script. A simple callback setup.
If I do this, at priority 3 the caller doesnt' get hungup, but
instead the line just keep ringing after callbback.agi is run. Why
is that?
exten =
Hi;
As you probably know, SER style of handling an
incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some
thing else
Is there any possibility of doingthe
aboveat "Asterisk Dial-plan"?
Regards
Mohammad
mohammad wrote:
Hi;
As you probably know, SER style of handling an incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some thing else
Is there any possibility of doing the above at Asterisk Dial-plan?
Just forward the call to Asterisk if it has a certain
Matt Riddell wrote:
mohammad wrote:
As you probably know, SER style of handling an incoming call is :
1) try to look-up it from registrar DB
2) if not found there, try to do some thing else
Is there any possibility of doing the above at Asterisk Dial-plan?
Just forward the call to Asterisk if
Matthew Simpson wrote:
Hello, I have a dial plan that tries to place a call over several
different outbound gateways, like this:
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,2,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,4,Dial(SIP/[EMAIL
Hello, I have a dial plan that tries to place a call over several different
outbound gateways, like this:
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,2,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,4,Dial(SIP/[EMAIL PROTECTED])
exten =
Matthew Simpson wrote:
What is the best way to handle this? Inserting +101 extensions with the
Hangup command ? Will that still properly signal 486 busy here back?
Should I be using Congestion instead of Hang up ?
What's the best way to handle it? Use my patch (unpublished) that stops
Dial()
Hi All,
I have an 800 number from Clearpath. Good folks, highly recommended.
I'd like to be able to use the 800 humber for DISA access as well as a
published number that I give to my customers. Does anyone here have any
example of dialplan logic that would handle normal 800 incomming calls,
then
'http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA'
This should help.
-Original Message-
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 9:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
I've been scratching my head for a while and I expect it is my mediocre
knowledge of Asterisk which is holding me back. If anyone can assist me with
some pointers I'd be grateful.
Basically, I've hooked up a Viking intercom at the front door. It hooks into
an fxs as a phone. Up till now
Il 01:20, mercoledì 29 dicembre 2004, Norman Zhang ha scritto:
Hi,
May I ask what does
exten = s,1,Answer
exten = s,2,ResponseTimeout(5)
exten = i,1,Playback(pbx-invalid)
s, t, i stands for? It says it is someexten but I still don't get it.
s: start is the extension invoked when there
First, please forgive me if this is a total newbie question, I've only
just begun to scratch the surface of asterisk.
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do
On Tue, 4 Jan 2005, Justin Richards wrote:
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do is set up
my dialplan to have an extension that offers up an outbound
Nice! that is exactly what I want!! much cleaner than the
callthrough plan (that I found two minutes after i posted my question)
I was trying to use that is posted on voip-info.org
Thanks!!
On Tue, 4 Jan 2005 17:24:36 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Tue, 4 Jan 2005,
Dear All,
I have two questions.
1.
How can I distribute equally the outgoing calls between channels in a
group?
I'd like to send outgoing calls on 2 ISDN GSM adapter.
I need to use the GSM subscriptions approximately equally.
2.
When all of the channels are busy in the first outgoing channel
I try to use my dialplan more general usable, ...
I defined the varable:
COUNTRYCODE=886 (Taiwan)
LOCALAREA=2 (Taipei)
To avoid any changes in the dialplan I would like to NOT allow to dial
the areacode I am in:
Is something like_9(N-${LOCALAREA}) available?
(above should give me
Hi,
May I ask what does
exten = s,1,Answer
exten = s,2,ResponseTimeout(5)
exten = i,1,Playback(pbx-invalid)
s, t, i stands for? It says it is someexten but I still don't get it.
Regards,
Norman Zhang
___
Asterisk-Users mailing list
PM
Subject: [Asterisk-Users] Dialplan variables
Hi,
May I ask what does
exten = s,1,Answer
exten = s,2,ResponseTimeout(5)
exten = i,1,Playback(pbx-invalid)
s, t, i stands for? It says it is someexten but I still don't get it.
Regards,
Norman Zhang
Norman Zhang wrote:
Hi,
May I ask what does
exten = s,1,Answer
exten = s,2,ResponseTimeout(5)
exten = i,1,Playback(pbx-invalid)
s, t, i stands for? It says it is someexten but I still don't get it.
Predefined Extension Names
Asterisk uses some extension names for special purposes:
* *i
-Original Message-
From: Chad Brown
Sent: Tuesday, December 21, 2004 8:02 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Flynn,
Yes, that makes sense. However, in my case I have incoming calls
arriving on an IAX channel from
Yes...Crystal.
Thanks Flynn
-Original Message-
From: el Flynn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 21, 2004 10:31 PM
To: Chad Brown
Subject: Re: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Chad Brown wrote:
Flynn,
You are being patient with me
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of
123-4567 and strip out the first four digits, and then dial whatever number
digits remain. If I only have three digits (000-999) and have a mix of
channels (ZAP, SIP, IAX2) could someone please point out how I can
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Samudra E. Haque
Sent: Sunday, December 19, 2004 12:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialplan selection
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten = _123,1,Dial(${X${EXTEN:4}},10)
Oops, that line should read:
exten = _123,1,Dial(${X${EXTEN:3}},10)
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What is the most efficient way to allow inbound callers to
dial internal users yet restrict them from outbound PSTN calls? Today I have a
basic greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I transfer the
Chad Brown wrote:
What is the most efficient way to allow inbound callers to dial internal
users yet restrict them from outbound PSTN calls? Today I have a basic
greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I
i would like to setup asterisk for any call that comes
in on a x100p is answered and is automatically
connected to a sip destination external to my
asterisk. what is the best dialplan to use? thanks in advance.
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--On Monday, December 06, 2004 2:48 PM -0800 Melbourne Lewis
[EMAIL PROTECTED] wrote:
i would like to setup asterisk for any call that comes
in on a x100p is answered and is automatically
connected to a sip destination external to my
asterisk. what is the best dialplan to use? thanks in
i have tried that. the sip party answers, but the zap
party continues to hear ringing. this is the conf.
zapata.conf
context=from-cell
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=1
busydetect=yes
busycount=4
channel=2
Hi - I have a zaptel card with 4 modules - 2 fxs and two fxo. I have two
phone lines coming into my house.
For now I want an incoming call to ring a phone here, and then if no
answer to ring another number (by calling out on the other line) for 15
seconds... then if no answer send to voicemail. It
Below is a section of my extensions.conf. As you can see I am trying to
route an incoming call to a remote phone number if it isn't picked up
within 15 seconds locally. Now this basically works.. the call is routed
to the remote number. But often it will still timeout and go to
voicemail.. as
Hello Asterisk,
is it possible to make an extensions that write a call file
(like a call back to the callerid) in
the outgoing directory WITHOUT using a perl AGI ?
--
Best regards,
Danny mailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures
-Original Message-
From: Danny Zak [mailto:[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 5:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Dialplan question
Hello Asterisk,
is it possible to make an extensions
I forgot to add a link to the system command:
http://www.voip-info.org/wiki-Asterisk+cmd+System
-Original Message-
From: Robert Jackson
Sent: Sunday, September 26, 2004 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialplan
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