the "correct" way to block these idiots so they
> don't even get this far.
>
> Thanks,
>
> Jerry
At this past year's AstriCon there was a series of security talks that
covered fail2ban and best practices. You can view the playlist of videos on
YouTube. The content s
o.
However, I would recommend you read "Asterisk the Definitive Guide". The
Future of Telephony is now an outdated version of the book and the name has
been changed to "the Definitive Guide." In the modern version of the book
there is installation instruction for both redhat-b
th
check-sync SIP NOTIFY
https://wiki.asterisk.org/wiki/display/DIGIUM/Provisioning#Provisioning-RemoteRestart
*Billy Chia*
Digium, Inc. | Product Marketing Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6099
*Check us out at*: www.digium.com &
touch function.
touch($callfilename,$time);
Kind Regards
Billy
On 5/3/12 7:28 PM, "Ashish Agarwal" wrote:
> How can I check how many lines are currently being used?
>
> On May 3, 2012 9:23 PM, "Duncan Turnbull" wrote:
>> Hi Ashish
>>
>>
kyoubye)
exten => t,n,Return
Inorder for the system to recognize invalid selections, I also changed
exten => i,1,Background(invalidentry)
exten => i,n,Goto(s,askuser)
To
exten => s,1,Background(invalidentry)
exten => s,n,Goto(s,askuser)
Thank you very much for the help.
Kind Regards
[s@macro-hangupcall:1] GotoIf("SIP/261-005c",
"1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/261-005c",
"TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] Go
Press 3 to Save the file
Note Each save file selection calls a different AGI file
E.g
exten => 1,1,Goto,timo|3552|9
exten => 2,1,Goto(3552,7) ; re-record message
exten => 3,1,Goto(4,1)
exten => 4,AGI(timorec.php)
Kind Regards
Billy
On 4/16/12 11:22 PM, "Dale Noll" wrot
7] GotoIf("SIP/440-004b",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/440-004b",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,13)
-- Executing [s@macro-hangupcall
Why not have multiple records in the sip.conf for the carrier.
For example, if your carrier was level3 then you'd do something like this:
sip.conf
[level3_729]
host=x.x.x.x
type=peer
insecure=very
context=whatever
disallow=all
allow=g729
[level3_ulaw]
host=x.x.x.x
type=peer
insecure=very
context=
FXS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of The VoIP
Connection
Sent: Wednesday, May 24, 2006 9:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP
FXS or FXO?
Michae
Use this:
exten => _1NXXNXX,2,Dial,IAX2/username:[EMAIL PROTECTED]/${EXTEN}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Sunday, May 14, 2006 2:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
Can someone please kill this guy's account?
Isn't there a Moderator on this list?
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, May 12, 2006 10:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] AT
On a full cone NAT, I have never been able to get the ATA to register
without a stun.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, May 02, 2006 1:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [
You will probably want to set a stun server in the 2100 if behind a nat. You
can use stun.fwdnet.net for testing. With that, you probably wont need to
port forward & it should work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Monday,
I’d set your box to DMZ on the
router & see if the problem exists first. If so, you probably forgot to
forward something.
Make sure that you forwarded both TCP
& UDP ports.
Billy P.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryn Nakaoka
Sent
I’m not sure about IAX, but in SIP… you can use rtcachefriends=yes
in the general section to accomplish this.
Don’t know about #2
Billy P.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Friday, April 28, 2006 8:40
PM
To: Asterisk
Users Mailing List
I’ve been using it for a few months
now. It works great. Needs some documentation but works really good.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scheda
Sent: Monday, April 24, 2006 10:32
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subje
Here is Grandstream's version of the spa-3000. I have used it and it works
great with asterisk.
http://grandstream.com/y-ht488.htm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tom
Sent: Saturday, April 22, 2006 8:35 AM
To: Asterisk Users Mailing List
Although there maybe a better way, this would work:
1. Add the IP's into your sip.conf and set qualify=yes.
2. Make your dialplan something like the following:
exten => _X.,1,Dial,SIP/[EMAIL PROTECTED]
exten => _X.,2,Hangup
exten => _X.,102,Dial,SIP/[EMAIL PROTECTED]
Interesting, I haven't set a hostname since I built the server almost a year
ago.
I wonder why only now would the problem arise.
William
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones
Sent: Wednesday, April 19, 2006 10:56 PM
To: Asterisk Users Mailing Li
Thanks, I found it though. I needed the
latest addons package. mysql CDR wasn’t there.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Wednesday, April 19, 2006
11:10 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
I deleted the modules directory, then ran
make and make install.
I then did “service asterisk start”
and asterisk –r but it says:
[EMAIL PROTECTED] asterisk-1.2.7.1]# asterisk
-r
Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)
[EMAIL PROTECTED] aste
Ok, I thought that was the case but I seem
to remember doing “make” then “make upgrade” in the
past. Is this no longer the way to do it or just another way?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Wednesday, April 19, 2006
9:51 PM
To:
List,
I wish to upgrade from 1.2.4 to 1.2.7.1
I have downloaded & unzipped the file but how do I
compile it?
Do I need to “make clean” then “make”
and “make upgrade”?
Or “make” then “make install”?
Thanks,
William
List,
The past few days the asterisk service on my server has
crashed several times. I have had it running for months and have made no
changes to it.
When it crashes, I am unable to make calls or gain access to
the CLI. The service has been stopped. If I try to start it again (serv
First of all, try sending it to the
asterisk-biz list.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Rich
Sent: Monday, April 17, 2006 10:53
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Don't
see my post
Hi Folks,
I have posted a c
Hi..
Have AAH set up with tdm card. 1 pstn line.
When incoming call initiated hard phone rings almost instantly.
Problem with outgoing calls from sipura spa 941, the call connects etc, but
is very slow to go out onto pstn.
There is a significant lag before the call at other end rings, perhaps as
i have AAH
connected to pstn via digium TDM01B
had been testing
it on telewest line (UK cable company) with very little issues.now moved to
a BT line and had several that i anticipated from infomation on this
list.the one that has caught me out is low volume from the caller via pstn.
u
oadvoice.com
fromuser=3109439023
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
canreinvite=no
authname=3109439023
[sip.broadvoice.com]
username=3109439023
user=3109439023
type=user
secret=mybvpassword
nat=yes
insecure=very
See this link: http://www.voip-info.org/tiki-index.php?page=NFAS
- Original Message -
From: "Billy Huddleston" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, October 12, 2005 12:57 AM
Subject: Re: [Asterisk
NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have
them in the same huntgroup at the CO.. All NFAS really does is free up a
extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you
pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the
I
Hi all
I'm trying to figure out what values are valid for the "insecure" option in a
realtime configuration table. The table field is 4 chars long and the actual
valid values for this is longer. Can I modify the field length or has this
changed? Below is where I looked, if I'm not looking in t
Does everyone have two processes running for mpg123? I always have them
when I'm running an idle Asterisk box. No calls going in or out and
nothing off hook. Is this normal? Thanks!
5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096
fpm-calm-ri
5015 ?S 0:00
Rob Scott wrote:
Is it possible to automatically set up a call between two external
lines?
I would like Asterisk is call a cellphone number, wait for it to answer,
and then call another cellphone, when that answers connect the two
together.
I assume it is possible but can someone point me how to
Billy Dunn wrote:
I am having one problem with the Polycom 600 phones. All phones on
the local network are fine and indicate presence to other phones
perfectly. One phone that is outside the network can see presence
indications of the other phones correctly, but that phone always shows
as
I am having one problem with the Polycom 600 phones. All phones on the
local network are fine and indicate presence to other phones perfectly.
One phone that is outside the network can see presence indications of
the other phones correctly, but that phone always shows as off the hook
to certa
Chris Mason (Lists) wrote:
Ideally it should talk to a local NTP server on your network, but I
have yet to see that work (but I'm only two weeks into Asterisk
too). Good luck.
Works for me, let me know if you need configs
Yes, I could use the configs on this. The phones are syncing to
[EMAIL PROTECTED] wrote:
There should be a NTP setting.
Setup Network Time Protocol.
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Louis-David Mitterrand wrote:
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
With the 1.5.2 firmware, have you managed to get one-touch message access
when
pressing the "Messages" button? It worked for me with 1.4.1 but no longer
with
1.5.2: I have to go th
Kristian Kielhofner wrote:
Billy Dunn wrote:
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to
ring-answer without the ring.
This is what I have in my sip.cfg file on the boot server:
voIpProt.SIP.alertI
dbruce wrote:
If you use the polycom provided config files, the default ring_answer class
is 4 and the auto_answer class is 3. So for your RANR alertinfo entry,
change the class to 3 and it will work as you expect. ie:
The correct ringtype entry should be:
Notice there is no timeout,
I'll give you an example. My staff does a lot of interviews with new
hires and salesmen. I do not personally sit in on the whole thing
(they can really drag out!), but I would like to be able to listen in
to make notes as needed while doing other work.
I had another idea whereas I could dial
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to ring-answer
without the ring.
This is what I have in my sip.cfg file on the boot server:
voIpProt.SIP.alertInfo.2.class="4"/>
voIpProt.SIP.alertInfo.3.class="5"/>
your problems into opportunities"
Thanks to everyone.
Billy
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A SIP phone *could* normally send its media stream directly from phone to
phone, if no transcoding is required, but when using Asterisk the media
stream will always pass through the server, causing a pottential
bottleneck.
So, why not use SER to register all the SIP phones, as it doesn't handle
t
7914 isn't supported with asterisk as of
yet.
- Original Message -
From:
Vasiliy Voropaev
To: [EMAIL PROTECTED]
Sent: Friday, October 15, 2004 6:32
PM
Subject: [Asterisk-Users] Cisco 7960 +
7914 - not worked
I have Cisco 7960 with 7914 operator console.
Problem was with asterisk.. Mark had made a change in chan_sip.c that
affected noncodec capabilities, it's been fixed.
Thanks,
Billy
- Original Message -
From: "Tenorio, Leandro" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion&
Sounds like you need to talk to polycom about a reduction in the
capabilities of thier phone after the upgrade and have them move the menu
option back..
- Original Message -
From: "Tor Setane" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Thursday, September
Cisco GW and DTMF problems
What version of IOS 're u using, and what's your dtmfmode in *?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisc
party. This affects DISA, prompts and
menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay
rtp-nte toggled in my dial peer..
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Administ
Dude, don't flam.. People don't use HTML capable E-Mail programs, or turn
off html for reason.. Like spam and web-bugs, and/or using classic email
programs like pine and mutt and linux.
Geez..
- Original Message -
From: Karl J. Vesterling
To: [EMAIL PROTECTED]
Sent: Sunday, August 01, 20
Asterisk doesn't support any form of G723 except with Pass through... You
might try G726 or G729
Thanks, Billy
- Original Message -
From: "Arnaud Pignard" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 23, 2004 12:45 PM
Subject: [Asterisk-Use
TN is one of the BEST states for BRI's.. Bellsouth messed up and had to
make some concessions to the PUC a long time ago.. You can get BRI
anywhere, and it's a flat fee.. typically $80-$90 per month Biz rate,
$35-45 Residential..
Tha
Do these work with PRI's as well? What's a ball park price on these?
Thanks, Billy
- Original Message -
From: "Robb Meredeth" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 09, 2004 8:53 PM
Subject: Re: [Asterisk-Users] T1 Hardware Echo C
I've been working with Mark today on fixing this very bug.. The patch
ProgramerTED did may have fixed it, but, I don't think it was the "right"
fix. We should have something done later today on this problem.
Thanks, Billy aka Connor
- Original Message -
From
alls on the PRI...
We found out that the 1-800 #'s go out our carriers Sonus switch (a VoIP
switch) which has 128ms Echo Can in it... Hmmm...
Thanks, Billy
- Original Message -
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, Jul
I've got the same problem NEAR end echo (We hear the echo on OUR side,
person on the PSTN never hears it..)
We're tyring to get our PRI carrier to run us through a echo can, or
re-write it through a switch they have which has built in echo cans...
Ugg..
Thanks, Billy
- Origin
Yes, I use it. Here's a sample extension of how to use it.
exten => 1234,1,Answer()
exten => 1234,2,MailboxExists(1234)
exten => 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no answer
goto voicemail
exten => 1234,4,Voicemail(b1234) ; send to voicemail if busy
exten => 1234,103,Dial(SIP/1
'local' target? What's that?
- Original Message -
From: "Matthew Asham" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
> You know, sleep deprivation cause people to do dumb things. The example
> I pasted was hastil
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading them.. along with all
the other folks..
Thanks, Billy
- Original Message -
From: "Mark Elkins" &l
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) an
hey, can you send me the tone?
- Original Message -
From:
Joe Antkowiak
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 4:30 PM
Subject: RE: [Asterisk-Users] Cisco 7940
Phones as paging system?
This is what we have
for this cu
SO, do you have a IDE CDROM?
- Original Message -
From: "Mark Elkins" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 07, 2004 4:13 PM
Subject: [Asterisk-Users] 729 licence on scsi
___
Asterisk-Users mailing list
[EMAIL PROTECTED
g711ulaw
- Original Message -
From: "Michael Shuler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 14, 2004 7:01 PM
Subject: [Asterisk-Users] Fax Over VoIP
> Anyone know what protocols support a fax machine i.e. g.729, g.711, etc?
>
> --
s worked up till this point.. What's going on?
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Administrator |
| Net-Express http://www.nxs.net |
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: "Marian Durkovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users]
der of codec preference for
> the gateway and see if that fixes your g729 phone and breaks the ulaw
phone
> at the same time.
>
> Alex.
>
> - Original Message -
> From: "Billy Huddleston" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent:
should be using the compatable codec... PLEASE help..
This is going to cause problems in our rollout.
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Administrator |
| Net-Express
unning 3.1, I've got several others, but this one is
the only one that does it...
Thanks, Billy
+------+
| Billy Huddleston Senior Systems Administrator |
| Net-Express http://www.
he Gateway or
ATA. Is their a way to eliminate this stutter without disabling re-invites?
This is very discontenting to our customers and employees...
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Adm
http://www.nxs.net/cisco_ata_186.htm
- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, February 08, 2004 12:40 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
> Could you share your 3.0.0 config?
>
> - Original Message
That's just it, I'm not doing anything.. Just normal use.. as far as I can
tell, they end up locking up with or without anyone using them as far as I
can tell..
Thanks, Billy
- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTE
Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or
anything, I am using re-invites. Pretty standard setup. When they lockup,
you can't ping them, or get to the http interface, and I even think the IVR
stops responding when you push the button.
Anyone had any problems with ATA's running 3.0 software locking up?
Thanks, Billy
+--+
| Billy HuddlestonSenior System Administrator |
| Net-Express http://www.nx
and with the HT-286 you get a Chinaman in a box! :)
- Original Message -
From: "Michael Koehler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 03, 2004 5:45 PM
Subject: Re: [Asterisk-Users] voip phones
> I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-28
won't have any problems.. It'll work fine for them.. What can we do to get
this to work like it should?
Thank, Billy
+------+
| Billy HuddlestonSenior System Administrator |
| Net-Express
. and MUST have it in the drive when you
bring Asterisk's up.
Thanks, Billy
- Original Message -
From: "Senad Jordanovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, January 21, 2004 5:28 PM
Subject: RE: [Asterisk-Users] G.729 Licenses from D
I've not had ANY problems using info OR rfc2833.. I did have problems using
inband. Try switching to it and see how it works.. I NEVER had a problem
with double digits, and, I believe that the reference to GS phones having
that problem with * was retracted.
Thanks, Billy
- Original Me
change dtmf to info on both * and in the handytone.
- Original Message -
From: "Senad Jordanovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 8:01 PM
Subject: [Asterisk-Users] Handytone 286 - calling out
> Hi,
>
> Just received recently released Grandst
loose username=ipphone9
Not needed.. the [109] is really the username
- Original Message -
From: "Anton Yurchenko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 24, 2003 11:42 AM
Subject: [Asterisk-Users] strange SIP authentication/authorization behaviour
> When I h
Use CIPE, It's a UDP based VPN solution.
- Original Message -
From: "Alastair Maw" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 20, 2003 1:37 PM
Subject: Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
> On 20/11/03 15:44, Chris Hirsch wrote:
> > Hey all..
how could you do this with sip and VOIP?
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 04, 2003 5:40 PM
Subject: Re: [Asterisk-Users] snatching calls
> On Tue, 2003-11-04 at 15:28, Shoval Tomer wrote:
> > Hi,
> >
> > O
;: Found
== Parsing '/etc/asterisk/voicemail.conf': Found
== Spawn extension (default, 9342199, 102) exited non-zero on
'SIP/-0810f210'
Anyone know what all this is about.. The Failed to write frame and the
'multipart/mixed;boundary=uniqueBoundary'
Thanks, Billy
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