(Author), Evan Henshaw-Plath (Author)
List Price: $49.99
Larry
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Presently I have _all_ 900 calls blocked in Asterisk 1.25
but today I had to call a parts vendor at a 972 number.
What are the safe 900 numbers - meaning the ones that are not sex
lines that change by the minute?
Larry
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Larry Alkoff wrote:
I would like to log a verbose statement in my 900/976 extens to a
special file called 'attacks'.
These are not standard messages like debug, notice, warning, error,
vebose or dtmf that could be logged to /var/log/asterisk/messages.
Does the 'verbose' in VERBOSE commands
with the
'verbose' in error messages?
I tried redirection of a VERBOSE statement - did not work.
Larry
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from
other devices such as a talking clock, driveway sensor or other home
automation devices like Stargate.
Larry
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exclude 900 calls.
My wife and I don't need them any more vbg
Hope this information will be helpful to someone else.
Larry
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Benny Amorsen wrote:
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA If it's not a security issue I might as well have all phones with
LA context=default in sip.conf even though voip-info specifically
LA warns against that. Wonder why?
Random SIP calls from the internet could end up
told this is very insecure.
How can I separate the outgoing extens?
When I create a context [outgoing] in extensions.conf with various
extens, they never get activated. How to I get them to dial etc?
Larry
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Benny Amorsen wrote:
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA I have a sip.conf with stanzas for sip phones that have
LA 'context=sip-incoming for some Grandstream phones and another
LA stanza for a Sipura SPA3000 with context=pstn-incoming.
LA Reviewing the code today, I was dismayed
= _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}
...
more exten lines to dial outside numbers here
[toll-access]
include = extensions
inclide = toll-trunks
Larry Alkoff wrote:
Benny Amorsen wrote:
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA I have a sip.conf with stanzas for sip phones that have
LA
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten = 91NXXNXX
more exten here
[toll-access]
include
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions
:/bin:/usr/bin:/usr/X11R6/bin) in new stack
Larry Alkoff wrote:
How can I access an environmental variable in Asterisk 1.2.5?
It should be possible according to:
http://www.voip-info.org/wiki/view/Asterisk+variables
which says:
Environment Variables
You may access unix environment
Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 08:04:23AM -0600, Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very
'myip is www.xxx.yyy.zzz'
exten = _4XX,n,VERBOSE(myip is ${ENV(MYIP)})
Why doesn't it work?
Larry
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looked at privacymanager and will try it if the above can't be
made to work.
Larry
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) to be changed to 'Internal'. I'd like to
avoid many lines of code so is there any way to do that with a wild card
or dial plan type?
Larry
Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make
to have just the line:
[410](grandstream)
Larry
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,SIPAddHeader(Call-Info: answer-after=0)
exten = **2,n,Page(${Two_Way_Intercom_List}|d)
exten = **2,n, Hangup
Two_Way_Intercom_List = SIP/420SIP/422/SIP/400SIP/413SIP/410
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Larry Alkoff wrote:
I am trying to page my Grandstream GXP-2000 phones
and keep getting the error message:
Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete
destination '' supplied.
How can I fix this error?
The two contexts below do either one-way paging or two-way
extensions each something like:
[412]
username=412
include=${grandstream}
Is there any syntax that would do this?
Larry
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asterisk
), and if you mean in the asterisk code,
there are logging examples all over the place.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Larry Alkoff
Sent: Tue 1/2/2007 11:22 AM
To: Asterisk-users; Austin-asterisk-users
Subject: [A*UG] How to show a debugging remark in a sip or extensions
I would like to show a remark that would show call progress
and appear on the CLI screen.
The remark should be in the code of a sip [channel] or extentions [context]
If I can't send my own remark, what little used 'show' command could I
insert in the code?
Can this be done?
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Larry Alkoff
.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Larry Alkoff
Sent: Tue 1/2/2007 11:22 AM
To: Asterisk-users; Austin-asterisk-users
Subject: [A*UG] How to show a debugging remark in a sip or extensions context?
I would like to show a remark that would show call progress
and appear
to cease
ringing as if the call was picked up)?
Finally, what should I put in dial plan 8 or elsewhere to send the call
to a context of my choice?
Larry
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in extensions.conf?
Or other?
Larry
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Larry Alkoff wrote:
I have a Sipura 3k connected to Asterisk 1.2.
All I want to do here is have incoming PSTN calls ring POTS phones
connected to the Sipura.
The web interface for the Sipura, on the PSTN line tab lists
VoIP User 1 Auth ID: asterisk
and
Dial Plan 8: (S0:66610)
How do I
:[EMAIL PROTECTED]
2006/11/23, Larry Alkoff [EMAIL PROTECTED]:
Problem: SPA3000 phone does not ring for incoming PSTN call although I
can dial out.
I set up my Sipura with the Voxilla Wizard which is pretty good but
leaves out some important details.
The Voxilla Wizard for Supura SPA3000 gave
address. You can see the last octet of the IP change.
Larry
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I would like to access my shell environment variable MYIP from within
sip.conf to put in externip.
I've tried some variations of syntax after reading The Future of
Telephony but it's not working yet.
Should it be
externip=${ENV{$MYIP}}
or some other syntax??
Larry
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analog
and SIP phones to ring at extension INRINGSEXT = 405 and would like to
see just how you do it.
Larry
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code
=dynamic
nat=no
port=5061
canreinvite=no
disallow=all
allow=alaw
allow=ulaw^M
allow=gsm
allow=g723.1^M
[EMAIL PROTECTED]
dtmfmode=rfc2833
On Sat, 18 Nov 2006, Larry Alkoff wrote:
Doug Crompton wrote:
Doug, please forgive me but I'm still having trouble understanding two
points from your last
username=sipurafxs1
secret=
context=default
context=from-pstn
; callerid=Doug Crompton 405
callerid=Larryy Alkoff 405
host=dynamic
nat=no
port=5061
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
[EMAIL PROTECTED]
dtmfmode=rfc2833
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Using
=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
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,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP
to
connect to server, connect FXS to FXO.
Steve
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Larry
,hangup
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gateway.
SIPaudio to SPA3k which converts it to POTSaudio.
Other calls are routed either to SIP extensions
or SIP provider. SPA3k is out of the picture.
Larry
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qualify=200
host=gw3.telasip.com
username=lalkoff
secret=xx
insecure=very
canreinvite=no
callerid=Larry Alkoff 5123011411
nat=yes
[200] ; Sipura Line 1 outbound to PSTN
type=friend
host=dynamic
context=home
secret=xxx
mailbox=200
dtmfmode=rfc2833
disallow=all
allow=ulaw
[201] ; Sipura forward
://www.grandstream.com/FAQ/Asterisk.htm
There's a PDF there that tells you (a) what settings to put on the
phone, and (b) how to configure Asterisk to sent the SIP header that
tells the phone to auto-answer.
Cheers,
Nic.
Please let me know if you get this working. I couldn't.
Larry
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=telasip-in in extensions.conf.
In extensions I have a [telasip-in] and [telasip-out] context.
Which if any of these are 'trunks'?
The Future of Telephony doesn't say much about trunks.
Larry
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Larry Alkoff N2LA - Austin TX
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/124SIP/125SIP/126SIP/127)
exten = s,2,hangup
From the sounds of it, this would probably work better if you setup call
queues, but the above will do what you are asking.
bp
On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote:
William I found and fixed the problem. Your comment gave me the kick
Original Message
Subject: How to use *411 using either last or first name?
Date: Thu, 31 Aug 2006 19:46:07 -0500
From: Larry Alkoff [EMAIL PROTECTED]
To: Austin-asterisk-users [EMAIL PROTECTED]
I read somewhere and put in my notes that
to make Asterisk accept either last
for 20 seconds
then go to priority 2.
exten =_879677[67],2,Dial(SIP/120SIP/122SIP/124)
bp
On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote:
Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120)
should be deleted?
Larry
William Piper wrote:
Sounds like you still have the old
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Larry Alkoff
Sent: Tuesday, August 29, 2006 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones
Color me puzzled. What part
= _352688,3,Dial,SIP/202SIP/214|20
Perhaps try sending everything in that context exactly as it is typed let
us look at it.
I'm pretty sure you have something configured incorrectly.
Thanks,
bp
On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote:
Sorry I was not clear Rushowr.
In the actual
This is a reply to a fairly old thread.
My EXTEN string is meant to ring 3 phones (will increase to 12) thus:
old: exten =_879677[67],1,Dial(SIP/120) ; works fine
new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124)
I edit extensions.conf to the new line above, type 'reload' into the
Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120)
should be deleted?
Larry
William Piper wrote:
Sounds like you still have the old exten still there.
Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120)
bp
On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote
and .=any number of digits.
I do not know if the underscore also interprets the * as something, or maybe it just gets stuck trying to figure out an expression
with no X nor .
Or this may not be an issue at all.
Larry Alkoff [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED
the
; extensions defined in variable Two_Way_Intercom_List which can be
; define as following:
Two_Way_Intercom_List = SIP/120SIP/122/SIP/100
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. Then use externhost=
instead of externip= in sip.conf . If you are using a
Linksys router like the WRT54G, it already has a
dyndns client which will update the dyndns servers
with your ip address everytime it changes.
Greg
--- Larry Alkoff [EMAIL PROTECTED] wrote:
As stated in the original
.
Greg
--- Larry Alkoff [EMAIL PROTECTED] wrote:
As stated in the original post, when I entter the IP
with an editor
directly into sip.conf calls work just fine but I am
looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is
possible
John Marvin wrote:
Larry Alkoff wrote:
As stated in the original post, when I entter the IP with an editor
directly into sip.conf calls work just fine but I am looking for a way
to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible for
Asterisk
the value into sip.conf programatically.
I could have just said how do I do this but wanted to show that I've
done my homework.
Thanks for any help.
Larry
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happen. However, each time I just use
externip=xx.xx.xx.xx the call works fine.
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cdr_manager.conf
cdr_custom.conf
cdr.conf
asterisk.conf
asterisk.adsi
alsa.conf
alarmreceiver.conf
agents.conf
adtranvofr.conf
adsi.conf
sip.conf
extensions.conf
extensions_additional.conf
voicemail.conf
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It _appears_ that the only way to create valid [context] is by a
context = line in sip.conf.
Is there another way to create a [new_context] in extensions.conf so I
can dial from it?
Right now most of my extens are in [default] and I'd like to avoid that.
Larry
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Larry Alkoff N2LA - Austin
Luigi Rizzo wrote:
On Thu, Mar 23, 2006 at 01:18:15PM -0600, Larry Alkoff wrote:
It _appears_ that the only way to create valid [context] is by a
context = line in sip.conf.
Is there another way to create a [new_context] in extensions.conf so I
can dial from it?
manually with an editor
BJ Weschke wrote:
On 3/23/06, Larry Alkoff [EMAIL PROTECTED] wrote:
It _appears_ that the only way to create valid [context] is by a
context = line in sip.conf.
Is there another way to create a [new_context] in extensions.conf so I
can dial from it?
Right now most of my extens are in [default
= 1234,1,Dial(SIP/1234)
exten = 1234,2,Hangup
Once you have that, just change the context = line to read context =
context_name and the phone will use that context for outgoing calls :)
Aaron
On Thu, 23 Mar 2006, Larry Alkoff wrote:
It _appears_ that the only way to create valid [context
be in the general section
and apply to all sip devices, or add per device and each can have its
own context
just adding to sip.conf will not result in a call
On Mar 23, 2006, at 1:18 PM, Larry Alkoff wrote:
It _appears_ that the only way to create valid [context] is by a
context = line
What do I have to do to dial an exten - with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
Larry
C F wrote:
Yes, you just create it.
On 3/23/06, Larry Alkoff [EMAIL PROTECTED] wrote:
It _appears_ that the only way to create valid [context
Yes I reload each time.
Larry
Aaron Daniel wrote:
Did you reload the dialplan in the CLI? I think it's extensions reload.
That'll refresh your settings... If that doesn't work, post your
dialplan so we can see what's going on :)
Aaron
On Thu, 23 Mar 2006, Larry Alkoff wrote:
That's how I
Hadley Rich wrote:
On Friday 24 March 2006 12:53, Larry Alkoff wrote:
What do I have to do to dial an exten - with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
There is a really good book available here[1] that will answer this and a lot
of other
the -5.0.3.EL part.
What should I put in where it says -5.0.13.EL ?? What is it about?
Larry
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, make progdocs
Shouldn't make install be _after_ make samples make progdocs?
5. Asterisk-sounds:
make install only
What about #4 - doesn't make install come after make samples/make progdocs?
Larry
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and Panasonic 2.4 cordless system would only
work over about 35 feet indoors - not enough for a large house.
Does anyone have any hands-on experience with DECT?
Larry
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for direct connection to CO line for emergencies). That way any single
(not double) line POTS phone can be safely plugged in as they are all on
line 1. Could this be done?
My mnemonic for telephone wire color coding is BOG Brown.
I'm interested in any comments on physical wiring.
Larry
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Larry
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