You don't mention if the phone is remote, or local. Although you do mention it
had a default user/pass. If the UI of the phone was/is accessible from the
I'net, the GUI does have the ability to place a call from it, that is one way
the calls could have been placed.
From:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, September 18, 2014 8:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling
[ewiel...@nyigc.com]
Sent: Thursday, March 13, 2014 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
Another option instead of 2 servers dedicated as PRI gateways is to use
AudioCodes Mediant 1000 or 2000 gateways. Either of them will also
failover to a backup proxy if the primary proxy (server) is offline.
Probably much cheaper than the kick ass box you plan to build + PRI
card(s).
I'm not
E911 does not follow the standard SIP RFC. That would be a good reason that
they couldn't/wouldn't do it. Now that I say that I should qualify it and say
NG911 (or ESINet) does not follow SIP RFC
http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your
county is not using
Sounds like you need disconnect supervision enabled somewhere.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Piszcz
Sent: Thursday, November 01, 2012 11:39 AM
To: asterisk-users@lists.digium.com
Or Audiocodes, or MediaTrix, or …
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani
Sent: Thursday, May 31, 2012 3:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; a...@avhan.com
Subject: Re:
This thread may interest you. Add a SSD and RAM and you're good to go!
http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200.
12460/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John
Voip.MS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Royce
Souther
Sent: Monday, March 05, 2012 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Link2VoIP going out of business! Now
I assume that solution was A2Billing?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 10, 2012 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Install Configure Fail2Ban then the host will be blocked from
connecting. And no, it's not new.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Saturday, November 26, 2011 6:55 AM
To:
sip.conf, look at externalip, externalhost, and localip.
From: Gopal krishnan
Sent: Wed 8/24/2011 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Integration with Android device
Hi,
I created a extension in Asterisk, the extension has
If this is what you need (fax/SIP/SIP Trunking/Vmail to email/Fax to email) and
are willing to run on real hardware, or a virtual machine (not an embedded
device), look in to PBX in a Flash along with IncrediblePBX/IncredibleFAX
addon. This setup will do everything you want, and then some. It
to see what OpenBTS is)
On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell te...@brummell.net
wrote:
Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure
other VTSP's) do.
From: A.H. Jos
Sent: Thu 7/28/2011 12:01 PM
To: Asterisk Users Mailing List
Google Voice will show your number no matter what, there was a problem with
abuse when they let you send the CID in the early days. Pretty sure there is
nothing you can do about it.
From: A.H. Jos
Sent: Thu 7/28/2011 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] hide
in the early days of GVoice?
On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.net wrote:
Google Voice will show your number no matter what, there was a problem with
abuse when they let you send the CID in the early days. Pretty sure there is
nothing you can do about it.
From
Really, since you sound like a novice in the Asterisk world, maybe
rolling your own solution isn't a good idea. Why not use an all-in-one
solution like PBX in a Flash?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
FreeBPX calls them Ring Groups, you can look in to that. Or you could
use a small ACD group.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
ghayyad
Sent: Saturday, July 02, 2011 12:58 PM
To:
-users] call paging interrupts call when using Mitel 5224
Thanks, that must mean it's not asterisk but the AGI/AMI software we use along
side it.
On Tue, Jun 21, 2011 at 7:27 PM, Terry Brummell te...@brummell.net wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Mitel 5224
How do you set them to Advanced SIP mode?
On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote:
My Mitel sets are all in Advanced SIP mode (I think that's what the call it),
have you done this? Once you change to Advanced SIP, you can't go back to
basic SIP
?
On Wed, Jun 22, 2011 at 8:47 AM, Terry Brummell te...@brummell.net wrote:
My Mitel sets are all in Advanced SIP mode (I think that's what the call it),
have you done this? Once you change to Advanced SIP, you can't go back to
basic SIP.
From: vip killa
Sent: Wed 6/22/2011 8:37 AM
Enhanced Mode and i
followed instructions in that PDF. would you be able to tell me what firmware
you are running?
On Wed, Jun 22, 2011 at 9:23 AM, Terry Brummell te...@brummell.net wrote:
http://edocs.mitel.com/UG/EN/SIP7.0_5212-5224_UG.pdf
Page 32
From: vip killa
Sent: Wed 6/22/2011 8:59 AM
the bootrom upgrade so i'm
still running 02.03.02.02
tested and call is still being interrupted when paging it...
are you running straight asterisk or is something else handling the dialplan
when you test?
On Wed, Jun 22, 2011 at 9:58 AM, Terry Brummell te...@brummell.net wrote:
R7.2.07.02.00.04
wrote:
Ahh then it makes sense, FreePBX checking to see if the line is in use, then
sending busy signal instead of interrupting the call
On Wed, Jun 22, 2011 at 10:13 AM, Terry Brummell te...@brummell.net wrote:
PIAF with * 1.8.3
My bootrom is 2.3.2.2 also.
From: vip killa
Sent: Wed 6/22
5224
Any chance you could send me (off list) you're example provisioning files
(without the SIP credentials and IPs of course)? I can't find them anywhere
online.
On Wed, Jun 22, 2011 at 11:21 AM, Terry Brummell te...@brummell.net wrote:
Yes.
From: vip killa
Sent: Wed 6/22/2011 10:56 AM
I have a 5224 and 5220's, I will try it tonight when I get home.
From: vip killa
Sent: Tue 6/21/2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Tuesday, June 21, 2011 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel
5224
Is
I didn't think it was possible if the endpoints, or Asterisk was behind a NAT.
Someone please correct me if I am wrong.
http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup
From: Sagbo Romaric
Sent: Sun 6/19/2011 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you
have a message. You press 1 to play and she just says First then gives you
options to delete, save etc. The message is in the INBOX as msg0001.wav
currently.
From: Alec Davis
Sent: Wed 6/15/2011 4:12 AM
To:
From: John Novack
Sent: Tue 6/14/2011 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
Robert Huddleston wrote:
Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Sent: Sunday, June 12, 2011 1:51 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] A question about Caller ID
Hi,
over anlog lines.
We use Jira at work. I hate it. Hope you have a better experience than
I've had!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell
Bryant
Sent: Wednesday, June 01, 2011 7:51 PM
To:
You are doing a CNAM lookup on that 202 number. Change the URL to a
number you know, and it will do a CNAM lookup on it. You can take your
tinfoil hat off now.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
From: Patrick Lists
Sent: Fri 5/27/2011 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] standalone PRI-to-SIP converter
On 05/27/2011 05:10 PM, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters.
For 2 different hosts. SIP/voxbone.com and SIP/4420
From: RSCL Mumbai
Sent: Thu 5/19/2011 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropping incompatible voice frame
Processor: Intel Dual Core Xeon 3.0GHz
- Host: CentOS 5.6 (64
From: Kelly Opal
Sent: Tue 5/3/2011 1:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dial from voicemail
Hi
Is it possible to dial from within voicemail to reach another extension.
I would like my customers to have a choice of dialing 1 to get my cell
phone while in
8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No messing
around with switches and cables an crap.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel
Sent: Saturday, April 30, 2011 10:09 AM
To: Asterisk
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, April 25, 2011 6:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PAP2T auto answer?
Hi all,
Is it possible to
http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory
From: Jonas Kellens
Sent: Mon 4/18/2011 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk unresponsive
Hello list,
I've got a whole lot of these in my debug
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Saturday, April 02, 2011 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration from
.
On Thu, Mar 31, 2011 at 4:05 AM, Gordon Henderson
mailto:gordon%2baster...@drogon.net wrote:
On Wed, 30 Mar 2011, Terry Brummell wrote:
Yah, sounds simple, how do you set it up to do this? Fail2Ban was
pretty easy, if it's that easy, why was F2B even created?
It's easy for me because I
I think you will find Fail2Ban the defacto standard.
From: vip killa
Sent: Wed 3/30/2011 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and fail2ban
so does anyone use fail2ban w/ asterisk or most people use sshguard?
--
elaborate on how you have iptables setup to work that
way?
On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson
gordon+aster...@drogon.net mailto:gordon%2baster...@drogon.net
wrote:
On Wed, 30 Mar 2011, Terry Brummell wrote:
I think you will find Fail2Ban the defacto standard.
I don't use fai2ban
VOIPo, CallCentric, F9, Voip.ms, CallWithUs to name a few (no particular
order, just what popped in my head)
From: asterisk-users-boun...@lists.digium.com on behalf of Brent A. Torrenga
Sent: Thu 3/3/2011 11:22 AM
To: asterisk-users@lists.digium.com
http://www.carrieraccessbilling.com/telecommunications-glossary-w.asp
From: asterisk-users-boun...@lists.digium.com on behalf of Gilles
Sent: Tue 3/1/2011 7:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [zapata.conf] What is wink?
Hello
When he says customers I am assuming he means remote customers. It
sounds like he is a reseller of telecom facilities to me. Which means
his customers most likely have ATA's with port 5060 forwarded to the
ATA, or they are direct on the I'net.
He has already set the ATA to only allow calls from
If you compare a working config with a non-working you will see something with
the answer type. I had that issue until I down rev'd. Look for something like
Ring Answer, I forget the exact details now.
From: asterisk-users-boun...@lists.digium.com on behalf
From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x
Hi Terry,
I did that, and did find a
From: asterisk-users-boun...@lists.digium.com on behalf of Mike
Sent: Thu 2/24/2011 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x
Hi Terry,
I did that, and did find a
Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation
for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All
are Asterisk based and very easy to set up.
From: asterisk-users-boun...@lists.digium.com
Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer to go with Elastix very easy to setup and maintain and reach UI rather
than freePBX
cheers
Dhaval
On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote:
Dean's link has references to Trixbox. TB
Yes, use a FXO device, like the AudioCodes MP-114. It is an external gateway
that will allow you to interface your PSTN lines to Asterisk via IP. There are
other brands out there but in my line of business we only use AudioCodes.
From:
Aastra Polycom because they can be configured using a TFTP server.
Great for large installations with centralized management.
Mitel 5215/5224 because they are dead simple to configure (via web gui)
and just plain work with no screwing around.
From:
Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know,
the Mitel's do not support tftp/http provisioning. I did just upgrade my
5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't
know what the phone is trying to do in that folder.
Anyway,
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