of somewhere else or simply
decides to forget where it came from as it does not appear to return
hence the System() call is never made.
I should point out I am using extensions.ael for my dialplan.
I personally have considered this behaviour to possibly be a bug.
Cheers,
Larry
for each of the nodes/peers,
no debug of each peer involved nor a trace of the packets hence no one
will have ideas!
Larry.
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the
whole packet and do the same with traffic between Asterisk UAC
Provider then use Wireshark and its telephony feature to analyse VoIP
calls, check the flows, you may discover the problem this way!
Larry.
M.
On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana
matteo.camp...@gmail.com
a different IP
address, if the latter I would suggest you check your configuration file
for bindings to specific IP addresses and make sure they match the new
machine.
Larry.
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On 14/04/2011 7:25 AM, Niccolò Belli wrote:
Il 13/04/2011 19:54, Larry Moore ha scritto:
That is because the remote endpoint, eutelia, will need to detect the
Fax Tones and send the T.38 ReINVITE to you, they may not have T.38
enabled.
Uhm... it's very unlikely.
I made a suggestion on how
On 14/04/2011 6:57 PM, Niccolò Belli wrote:
Il 14/04/2011 12:25, Larry Moore ha scritto:
I made a suggestion on how you could check this i.e. have your incoming
call go directly to the fax extension, my 1.8.3.2 installation
immediately negotiates a T.38 connection in this sceanrio, of course I
On 14/04/2011 10:16 PM, Niccolò Belli wrote:
Il 14/04/2011 14:34, Larry Moore ha scritto:
allow=alaw,g729 ; alaw required for T.30 facsimile if T.38 fails to
I didn't understand the point. If you enable both alaw and g729 it will
simply use the preferred one: if it's g729 tone based detection
expect you will then see T.38 re-invite come from Asterisk.
What is in your configuration for 159?
Larry.
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On 6/04/2011 4:27 AM, isr...@gmail.com wrote:
Ok thanks I found the problem
Your welcome, can I take it that you captured the packets, you then
viewed them in Wireshark and that is how you discovered the issue?
Larry
it has gathered the information it wants you will be able to select
one ore more sessions, once selected click on the Flow button, all will
now be clear!
Larry.
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and traces of the sessions.
If you turn off all T.38 options in Asterisk and on the SPA you should
still be able to make a transmissions using the G711 codecs.
Can you confirm you are able to send a facsimile from your device using
a PSTN line?
Larry
: no
Message Waiting: noAccept Media Loopback Request: automatic
Media Loopback Mode: sourceMedia Loopback Type: media
Larry.
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Jeremy Kister wrote:
On 12/30/2010 9:59 PM, Larry Wimble wrote:
I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small
VPS (512mb standard, 512mb burst). I note that the asterisk process
is using about 209mb of memory just doing nothing (not configured to do
anything yet)
I'm
?
Thanks,
Larry
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Hello,
What do you get with ipconfig from your clients?
What do you get with nslookup of a client or server?
What do you get with tracert to your server from a client?
Can you access the internet from a client? Are you isolated as a private
network?
Thanks
Larry
-Original Message
Larry
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why and Is this a bug ?
Thanks!
Larry
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and so forth. It just set the phone number and extension in the * for
inter used . Could you tell me the reason, and how I could get the number of
the fxs?
Thanks
Larry
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and so forth. It just set the phone number and extension in the * for
inter used . Could you tell me the reason, and how I could get the number of
the fxs?
Thanks
Larry
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the
number of the fxo which is used as a common analog phone?
Thanks
Larry
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HI
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext = 700
parkpos
HI
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext = 700
parkpos
-
are they?!?!
Can someone in this good group please help me with some advice as to who can
provide affordable and reliable international toll free service for a better
price than ATT?
Thanks in advance,
Larry Costigan
Food Donation Connection
(Asterisk fan and ABE user
to improve the voice quality is
greatly appreciated.
Sincerely,
Larry Costigan
Food Donation Connection
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attepting this?
Another option might be to setup Asterisk to interface with MySQL and then
work out the details of exchanging data between MySQL and SQL Server...
Any and all help is greatly appreciated!!
-Larry
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... And welcome to Asterisk!!
:-)
-Larry
On 8/16/07, Barry L. Kline [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Bill Andersen wrote:
[snip]
Would I be better off starting with:
a) Plain old asterisk from asterisk.org?
(tutorial suggestions?)
b
(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
to [toll-access]?
Also I don't understand the 'doubling' of [extensions] by including it
in another context.
I'm probably missing something here. Can you help me understand this
better?
Larry
Eric ManxPower Wieling wrote:
Put your phones in the context=toll-access in sip.conf
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
I recently read about the following new technologies from Digium. Has
anyone tried the new HPEC or knows when it will be available?
TDM800P and HPEC
The TDM800P is an 8-port analog telephony interface card, so it fills the
gap between Digium's 4-port and 24-port cards. Analog phones and POTS
$MYIP
both work fine.
Larry
Ioan Indreias wrote:
Hello Larry,
Probably your variable (MYIP) is not accessible to asterisk process
environment.
Test it with ${ENV(PATH)} and you will have a result there
exten = s,n,Set(test=${ENV(PATH)})
-- Executing Set(IAX2/test_iax,
test=/sbin:/usr/sbin
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll never be able to read the
MYIP variable from Asterisk.
Larry
'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
Thanks very much Chris.
I found usage for NoOp and verbose in Future of Telephony Appendix C
and it looks like they will do exactly what I need.
Larry
Chris Tooley wrote:
If you mean in the dialplan, you can use NoOp or verbose (verbose being
something that will get logged too
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?
--
Larry Alkoff
Chris how would I use 'verbose' in a dialplan context?
A sample line?
Larry
Chris Tooley wrote:
If you mean in the dialplan, you can use NoOp or verbose (verbose being
something that will get logged too), and if you mean in the asterisk code,
there are logging examples all over the place
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
Fran when you say specify the next hop do you mean the S0 line be an
extension in sip.conf or a context in extensions.conf?
Or should the line simply be tacked on to my [default] context?
Larry
Fran Oliveira wrote:
I think it is wrong. You should specify the next hop with some like this
S0
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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with this.
The Wizard was nice enough to give detailed settings for sip.conf and
extensions.conf but nothing about to handle Dial Plan 8 except You'll
need to enter the extension you wish to forward all incoming PSTN calls
to on your Asterisk server. I don't understand how to do that.
Larry
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
Thanks very much for your sipurafxs1.
The problem has been that incoming POTS calls are swallowed up after the
first ring or so if the pstn line is connected to Sipura.
I'll try this and let you know.
Larry
Doug Crompton wrote:
This is my spa3k fxs port sip.conf params. This uses
.
My guess is that I should modify username=spa3k-pstn-in
to conform to your sipurafxs1 but I'd like a reality check as I've lost
the overall picture.
Larry
Doug Crompton wrote:
This is my spa3k fxs port sip.conf params. This uses the default context
in my extensions.conf
What are you
Doug tThis is my latest sip.conf and extensions.conf with your
sipuraxfs1 renamed to spa3k-pstn-in as it's called in my Sipura.
The non-Sipura parts have been mostly snipped.
I still have the problem that an incoming POTS call is swallowed up
after the 2nd ring.
Larry
[EMAIL PROTECTED
extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set
be set to
“none”. This case also belongs to call type #7 and the voice path is (1)
(2) (4) (6) (7).
Of course, I'm having a lot of trouble reading this complex manual g
Larry
Bob Chiodini wrote:
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in:
http://www.sipura.com/Documents
,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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up.
Does that sound like a situation that would be helped by call queues?
Larry
William Piper wrote:
Sure, do something like this:
[telasip-in]
exten = _512879677[67],1,macro(callgroup,s,1)
exten = _879677[67],1,macro(callgroup,s,1)
[macro-callgroup]
exten = s,1,Dial(SIP/120SIP/121SIP/122SIP
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
Sorry I was not clear William.
In the actual code, the exten marked 'old' is commented out and only
'new' is active. Then I reload. But only the single 120 instrument rings.
Larry
William Piper wrote:
The whole thing.
Both (old and new) have the same exten and the same priority, you can't
Sorry I was not clear Rushowr.
In the actual extensions.conf as used, the 'old' line is commented out
so only 'new' is active. Then I reload. However, only the single 120
line rings instead of all.
Larry
Rushowr wrote:
Then entire OLD extension must be removed so the new one will match
-in]
and other contexts to call ring groups for extension intercomming.
Is there some kind of macro I could have to replace the instances of:
(SIP/120SIP/122SIP/124)
I have not yet written or read up on macros.
Larry
William Piper wrote:
I don't know then, I do the same exact thing:
exten
' into the
CLI, see the new line with 'show dialplan' and actually see the new line
above, but when I dial the DID 879-6777 it rings on extension 120 only.
Have I missed a step?
Larry
Jonathan k. Creasy wrote:
EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE)
From: [EMAIL
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
)
into extensions.conf and, under CLI, issued reload.
Is that the correct place?
Larry
Steven wrote:
I do not know if this breaks anything or not the way you have it, but you
should not have the underscore before the extension.
The underscore means that the following is an expression, where X=any single
digit
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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containing the IP information in
the form of ${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf
That's a very nice idea Greg. I'm not sure that my Asterisk 1.2 has the
externhost= function but it would solve my problem.
I have a dyndns.org account already that reports my externip.
Larry
Greg Delgado wrote:
The easiest way is to register for free dynamic DNS
service at www.dyndns.com
Thank you Greg and RR.
externhost=myhost.dyndns.org works perfectly so figuring out how to
access a shell variable from within the CLI is no longer necessary -
although it would be nice to know!
externhost works in 1.20 onwards.
Thanks for finding the solution.
Larry
Greg Delgado wrote
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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RR wrote:
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis
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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to
find them.
Outgoing CNAM is a different beast however. They can't take it via IE.
You need to get it into their database (they have two)
and have them push it to all the other telco's.
-larry
Message: 10
Date: Mon, 10 Apr 2006 22:42:35 -0400
From: Andres [EMAIL PROTECTED]
Subject: Re
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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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
That's how I _thought_ it worked but extens in such a created
[context_name] are not seen or used by Asterisk to dial out.
There is something missing.
Larry
Aaron Daniel wrote:
Yes.
Just create a context that you want the phones to dial from in
extensions.conf.
[context_name]
exten
in sip.conf.
Are you talking about a [users] context? Another thing I don't
understand about.
Larry
Jerry Jones wrote:
You need to create [new_context] in extensions.conf
then add the context=new_context to sip.conf so calls from from the sip
devices know which context to use. this can
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
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Asterisk-Users mailing list
part of the time. But from my experiance
DTMF is not handled correctly in asterisk if you use any
gateway other then asterisk.
IE: you use a cisco or TNT as your gateway to/from the PSTN via SIP and
asterisk to talk to the 2500 type phones.
-larry
Message: 1
Date: Thu, 16 Mar 2006 12:36:45
, 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
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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I need to detect when a fax machine answeres an outgoing call. NV_FaxDetect
and the zaptel fax detect seem to only work in calls originated FROM a fax
machine, not for calls ANSWERED by a fax.
Thanks
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Larry Host
NuWorld Telecom, Ltd.
858-334-9355 Cell
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[EMAIL
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