Anyone have any recent experience with openfire and asterisk
integration, perhaps with the spark IM client? About to dive into this
and would appreciate any advice on gotchas.
Cheers,
j
--
_
-- Bandwidth and Colocation Provi
On 11/06/2017 12:34 PM, Joshua Colp wrote:
On Mon, Nov 6, 2017, at 02:14 PM, Saint Michael wrote:
Asterisk is unique in terms that we can create new applications that talk
to databases and generate any logic whatsoever. Asterisk is a development
environment for anything telecom, not a PBX. I bel
Hi,
We have been using Voxbone for some time for origination, and they now
offer E911 services. We are trying to set this up and having trouble
meeting their authentication requirements.
I setup a peer as I normally would, with user/pass as they supplied
("lacoursj", "pass"), but my calls a
gt;)
ofcourse for each customer you will need to provide his own did.
On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere <mailto:j...@stratustalk.com>> wrote:
Hi,
We have been using Voxbone for some time for origination, and they
now offer E911 services. We are trying to
05/08/2018 02:54 PM, Khalil Khamlichi wrote:
try adding a + sign for the number
same => n,Set(CALLERID(all)=17864089672 <+17864089672>)
On Tue, May 8, 2018, 8:51 PM Jeff LaCoursiere <mailto:j...@stratustalk.com>> wrote:
I *am* doing that, as I assumed it would b
I've been struggling for a few weeks now with the local telco trying to
bring up a trunk that has been down for a year (hurricanes in the
caribbean). Box is a Dell R710, 16G RAM, Ubuntu 14.04.5 LTS, Dahdi
2.10.2-rc1, asterisk 13.23.1. Xorcom Astribank w/ one T1/E1/PRI module,
plugged into a
one from Digium can pitch in. I suppose I should have some kind of
support with the G100... have never tried to actually call Digium before.
Cheers,
Jeff LaCoursiere
--
_
-- Bandwidth and Colocation Provided by http://www.
one from Digium can pitch in. I suppose I should have some kind of
support with the G100... have never tried to actually call Digium before.
Cheers,
Jeff LaCoursiere
--
_
-- Bandwidth and Colocation Provided by http://www.
tream.
*Nick Olsen*
Network Engineer
Office: 321-408-5000
Mobile: 321-794-0763
--------
*From*: Jeff LaCoursiere
*Sent*: 2/15/19 1:12 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject*: [asterisk-users] Di
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815
Our provisioning servers listen on a high numbered port. We generally
don't have any issues with scanning...
Cheers,
j
On 6/18/19 7:18 AM, John Runyon wrote:
Just to jump in on this, this just started happening to our system a
couple days ago. (To the tune of 3GB of webserver access logs y
ce '0x7fe17426e7c8'
So my main question is, what would cause a sixteen second delay before
the codec could be decided?
This is Asterisk 13.25.0 on the customer Amazon instance... the "ast01"
peer is ours also - one of our external gateways, also running 13.25.0.
Thanks
,
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*
In a few weeks? FIrst I have heard of this, and your legitimacy is
strained by a gmail address.
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
W
See also:
https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratust
I'm pretty sure that means your are using a non-color capable terminal,
or your termtype variable is incorrect. What are you using for a
terminal emulator?
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email
the remote shell (is it remote?) can't
determine your termtype. This is pretty ancient code.
j
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk
I work from a similar setup. I ssh'ed to my personal PBX from an xterm
window on an Ubuntu 16 workstation, your prompt seems to work:
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <
Isn't the MySQL stuff deprecated in favor of odbc? You may be barking up
the wrong tree if you plan to make source changes.
j
On Sun, Jun 7, 2020, 1:55 AM Fourhundred Thecat <400the...@gmx.ch> wrote:
> > On 2020-06-06 10:38, Antony Stone wrote:
> > On Saturday 06 June 2020 at 09:18:11, Fourhun
ng them way late, in which case the issue is
actually your hardware.
Cheers,
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://ww
On 6/15/20 2:19 PM, Luca Bertoncello wrote:
Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere:
Hi Jeff,
We are working on a product to analyze pcap files of VoIP calls. So far
it does a reasonable job of analyzing the frequency distribution of
packets in both directions, pointing out which
On 6/16/20 1:18 AM, Luca Bertoncello wrote:
Am 15.06.2020 23:15, schrieb Jeff LaCoursiere:
Hi again,
just a question, to be sure...
sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &
eth0 is my DSL interface and eth1 my phone interface?
t since I'll change my contract
at 18.06., I'll wait some days. Then I'll have a "business" contract,
and I hope I don't must speak with someone that can just say "you have
to reboot your Fritzbox. What? You don't have a Fritzbox? That's not
pos
interfaces involved?
Always looking for real-world data to improve our tools :)
Cheers,
--
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*h
ou were presenting it. Who is going to base their
business on some list guy with a gmail address?
--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
--
_
-- Bandwidth and Colocation Provided by http://www.api
you are looking for...
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On 10/29/20 9:05 PM, David Cunningham wrote:
Hi Dovid,
We can change the SDP in Kamailio, but Asterisk will still send its
RTP from its default address. The remote end is strict about accepting
RTP from the specified source and
you are looking for...
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On 10/29/20 7:42 PM, David Cunningham wrote:
Hello,
Does anyone know a way with chan_sip to tell Asterisk to use a
specific IP address for its end of the communication for a specific
device? Something like:
[device]
typ
you are looking for...
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On 10/29/20 7:42 PM, David Cunningham wrote:
Hello,
Does anyone know a way with chan_sip to tell Asterisk to use a
specific IP address for its end of the communication for a specific
device? Something like:
[device]
typ
This machine I visited yesterday in our data center... it is running
Ubuntu 14... I would say this is a pretty stable platform :)
Cheers,
j
On 12/9/20 5:00 AM, Dmitry Melekhov wrote:
09.12.2020 13:20, Frank Vanoni пишет:
On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote:
what is
Chemin de BADASSAC
34510 FLORENSAC
Cel : +33/0 638 42 91 93
http://www.facerias.org
Le 2020-12-09 14:06, Dmitry Melekhov a écrit :
09.12.2020 16:52, Jeff LaCoursiere пишет:
This machine I visited yesterday in our data center... it is running
Ubuntu 14... I would say this is a pretty stabl
hy run asterisk?
Am I missing something?
Cheers,
--
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*
are rejected, I can just get him to
sign them for me? If I were him I would get a bunch of lawyers ready
for when he becomes responsible for what they end up doing. Isn't that
the whole idea?
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On 1/25/21 7:44 AM, Joshua C. Colp wrote:
On Sun,
A 40Kb limit seems a bit draconian these days. I simply attached a
small pic to illustrate a point. May I vote to up the limit? 100K?
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
Forwarded Message
Delivered-To: j...@stratustalk.com
Received: by 2002:a05:6602:44b:0:0:0
On 1/25/21 12:12 PM, Steve Edwards wrote:
On Mon, 25 Jan 2021, Jeff LaCoursiere wrote:
So how does this guy get around it? It sounds to me like he is
offering to sign calls for whoever, which IMO totally defeats the
purpose.
IIRC, back when he first started hawking his solution, he
I'm planning to make a big post in
a week or so with all I have learned, hopefully will help others unsure
where we stand. June is coming up quick!
Cheers,
--
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:
them.
Just to verify, this is the same script running over and over with the
same parameter.
Any ideas/suggestions as of what can be happening?
Thanks,
Alex
--
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *j...@
ngs that may do so.
Basically we can't do LCR anymore. Outbound calls are locked to the
provider that gave us the DID. I'm not sure that's really a bad thing,
its less headache than for us to try to become a signing authority.
I think the whole thing is still very fluid. Didn'
Hi Alex,
Are they doing anything on inbound for you, and have you made any
decisions about how you will display the tag to your customers? I have
been focusing on the outbound piece of this, just starting to think
about what to do with the incoming data...
Cheers,
Jeff LaCoursiere
on
the carrier that "owns" them, and not worry about this.
Jeff LaCoursiere
StratusTalk, Inc.
On 3/11/21 8:12 PM, d...@donkelly.biz wrote:
You said it in your first post when you said “I reallt don’t
understand.” You don’t understand the business that these people are
in. A few pe
ialplan variable and in the CDRs.
Jeff LaCoursiere
StratusTalk, Inc.
On 3/11/21 6:21 PM, Alexander Perkins wrote:
Hi Jeff. What exactly do you mean by the 'inbound piece'? I've spent
quite a lot of time with the folks at TILTX understanding the
framework; but I am not exactly
o DNS resolution
of Asterisk's hostname.
Try to add your hostname to /etc/hosts and check if it's better.
Regards,
Jean
--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
--
_
-- Bandwidth
Haven't tried this, but try piping through 'strings'
j
On Wed, Aug 3, 2022 at 6:13 PM Carlos Chavez wrote:
> The "-n" option only works on startup and cannot be used when Asterisk
> is already running (I tried and I get an error). We are using version
> 18.12.1. The output I want to captu
Hi all,
Curious if the github user "mlan" is on this list? Could you please
contact me off list if so, I was hoping to reference your work in a talk
at Astricon next week, and... I don't know how to contact github users lol.
Cheers,
--
Jeff LaCoursiere
StratusTalk, Inc.
70
Ah, great advice, thanks!
j
On Tue, Feb 7, 2023 at 10:09 PM John Runyon wrote:
> If you clone one of their repo's you can see their email address in the
> commit log...
>
> On Tue, 7 Feb 2023 at 16:56, Jeff LaCoursiere
> wrote:
>
>> Hi all,
>>
>> Cur
ages on your
Debian
system, but if not, then there is no reason to pay any attention at
all to
anything to do with Alembic, Realtime, SQL etc.
Antony.
--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
--
__
Howdy,
Has anyone worked on a Mitel-2000 emulation for PMS integration (Hotel
mgmt systems)? Hoping to get my hands on the protocol definition
(RS-232!!) for check-in/check-out/housekeeping/CDR, but if someone has
already done I would totally buy it.
Cheers,
--
Jeff LaCoursiere
fully
explain what you are trying to accomplish, and some folks here can try
to steer you towards a workable solution. It almost sounds... nefarious.
Cheers,
--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
81
On 6/26/23 5:19 PM, Jeff LaCoursiere wrote:
On 6/26/23 9:00 AM, Joshua C. Colp wrote:
On Mon, Jun 26, 2023 at 10:57 AM TTT wrote:
I am connecting to the ARI with subscribe all, so I can see
channels being created. I now want to extract a variety of
header variables (at the moment
for a call in progress, but that
sounds like a huge project. I’ve got to demonstrate something by end
of week.
*From:*asterisk-users [mailto:asterisk-users-boun...@lists.digium.com]
*On Behalf Of *Jeff LaCoursiere
*Sent:* Monday, June 26, 2023 6:20 PM
*To:* asterisk-users@lists.digium.com
Bandwidth.com, although there are minimums to meet.
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On Fri, Aug 18, 2023 at 7:52 AM TTT wrote:
> Check out Twilio
>
>
>
>
>
> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Behalf Of *Federico
&g
D number I dialed. So why is the new box
immediately answering the call before it gets all the ANI digits? I've
poured over all Sangoma's docs online for debugging RBS lines - I'm not
having any transmission e
Lumenvox is actually both... but hard to justify for TTS given all the
freebies...
On 01/10/2014 02:50 PM, Todd R. wrote:
Actually, scratch that.. Luminvox is not text to speech it's speech
recognition software. Got this mixed up and turned around :-) Anyhow,
see the link I posted earlier, i
On 01/19/2014 08:40 AM, Steve Murphy wrote:
Here's another idea! How about changing your port from 5060 to something
different, maybe 7067 or some other number that is not popularly being
used?
You'll provision your phones to use this port, and the scanners will not
find you. Seems a much simp
On 03/24/2014 05:50 PM, Thorolf Godawa wrote:
Hi,
I'm installing Hylafax on my Asterisk system. From what I've read, I
can either use IAXModem or T38Modem to provide the virtual fax device.
we are using T38modem, it was a long way to get it stable, but finally
it works quite good with 10 para
On 03/25/2014 10:13 AM, Steven Howes wrote:
On 25 Mar 2014, at 15:00, Jeff LaCoursiere wrote:
On 03/24/2014 05:50 PM, Thorolf Godawa wrote:
But your carrier has to support T38, when we began to evaluate this some
years ago, this was not true for all.
Would you share the provider you are
On 03/26/2014 10:05 AM, Michelle Dupuis wrote:
I see a lot of attempts by hackers to call
00972595301123or 011972595115207 or variations but that same 972595 is
often present.
Can someone break down that dial string with an explanation? The 011
look like an overseas call (from Americas),
Hello,
I am having odd issues with a new PRI based installation. Outbound
calls work for all numbers except those that terminate at Sprint! The
telco is new to PRI (this is in the Caribbean) and say that Sprint is
rejecting the calls, and asked for our PRI settings so they can work
with thei
isk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, August 19, 2014 5:00 PM
To: asterisk-users@lists.digium.com >> Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Way to dump PRI
Hello,
I wrote earlier today about a new PRI installation in the Caribbean,
where all outbound calls are functioning fine *except* calls to Sprint
phone numbers, which get rejected immediately as "busy".
The telco has been working with their switch manufacturer and took the
output of "pri sh
On 08/20/2014 07:58 AM, Scott L. Lykens wrote:
On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere <mailto:j...@jeff.net>> wrote:
I wrote earlier today about a new PRI installation in the Caribbean,
where all outbound calls are functioning fine *except* calls to
Sprint phone numbers,
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff
LaCoursiere
*Sent:* Wednesday, August 20, 2014 2:41 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings
On 08/20/2014 07:58 AM, Scott L. Lykens wrote:
On
ssary.
--Don
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff
LaCoursiere
*Sent:* Wednesday, August 20, 2014 10:03 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings
What about the
On 08/20/2014 12:04 PM, Andres wrote:
On 8/20/14, 11:28 AM, Steve Totaro wrote:
PRI intense debug should show all you need to fix this.
Right, the sooner you post this debug here the sooner we can help.
Otherwise its just guesswork.
On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere
Ok, here is an intense debug trace. I've replaced the phone numbers
to protect the innocent. The smoking gun seems to be this:
Ext: 1 Cause: Destination out of order (27)
Though I have no idea why... calling the same destination from my cell
phone works fine. We only send seven digits fo
number in your dial?
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff
LaCoursiere
*Sent:* Wednesday, August 20, 2014 5:29 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings
On 08/20
On 08/26/2014 09:55 AM, Doug Lytle wrote:
What I found curious was the caller's name was Asterisk
On our systems, if I don't assign a CID number to an inbound call that is
blocking it's CID, the default shown on the Polycom phones is Asterisk. I've
set it up that any inbound call with no CID
On 09/02/2014 03:14 PM, Administrator TOOTAI wrote:
Le 02/09/2014 20:18, Khalid Touati a écrit :
so it seems Asterisk Versions does not support video I guess
Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11
with GrandStream phones (H263, H263+ and H264). Works perfectly
Don't forget "videosupport=yes" in sip.conf.
j
On 09/02/2014 03:52 PM, Eric Wieling wrote:
A co-worker was doing video, I dislike video. The phones were Polycom VVX, The
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP
Settings / Video section we have Video: Enabled,
On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
So, once I have the audio in the database, how can I play it?
Creating temporary files seems so tacky.
Is there another way to playback
On 09/23/2014 10:53 PM, Don Kelly wrote:
On Tue, 23 Sep 2014, Steve Edwards wrote:
On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
On Tue, 23 Sep 2014, Don Kelly wrote:
I'm
On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
Hi
Will the presentations made at Astricom 2014 be made public as recorded videos ?
thanks
Bogdan
I'll second the request for that, and also ask if the sessions on
Kamailio will be similarly available.
Cheers,
j
--
__
On 02/25/2015 09:28 AM, Steve Edwards wrote:
On Wed, 25 Feb 2015, A J Stiles wrote:
The limiting factor with a switch carrying IP telephony traffic is
not bandwidth, but routing table entries; and even cheap switches
nowadays will usually take 1024 entries, if not 4096.
Are you referring to
Why use Amazon? With that kind of load I would want dedicated servers.
Call Rackspace or Softlayer.
j
On 03/06/2015 11:59 AM, Amit Patkar wrote:
Hi
I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 +
g.711) and full reco
instance.
*Thanks & Regards,*
Amit Patkar
On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote:
Why use Amazon? With that kind of load I would want dedicated
servers. Call Rackspace or Softlayer.
j
On 03/06/2015 11:59 AM, Amit Patkar wrote:
Hi
I plan to host Asterisk instances on AWS/
s ssd on all the virtual machines. Uptime is good.
Jai Rangi
Www.didforsale.com <http://Www.didforsale.com>
www.cebodtelecom.com <http://www.cebodtelecom.com>
www.cebod.com <http://www.cebod.com>
On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere <mailto:j...@jeff.net>> wrote:
Only slightly asterisk related I suppose, but hoping someone has
attempted this...
I have an old installation with a bunch of IP501s, and one died. I
replaced it with an IP450, and the user sorely misses his DND button. I
hated those DND buttons anyway, as I couldn't control them centrally
Hey Don,
How are you? I may be heading your way in the next month or so. Have to
meet with a guy in Eden Prairie, and stop off at my
brother/sisterm-in-law's as well.
Got a question for you - with TBCT, who pays for the call once it is
transferred? Still me as the owner of the trunk?
Le
My, how embarrassing. I of course meant that as a personal message to
Don. But if anyone else knows the answer, I'm interested! lol
Cheers,
j
On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote:
Hey Don,
How are you? I may be heading your way in the next month or so. Have
to meet w
Hello,
I am wondering if asterisk does anything at all to RTP packets passed
from channel to channel if no transcoding is involved? Can I assume that
the packet that left phone A, arrived at the asterisk server, was copied
to phone B's channel and eventually arrived at phone B had exactly (by
On 03/24/2015 04:28 PM, Richard Mudgett wrote:
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <mailto:j...@jeff.net>> wrote:
Hello,
I am wondering if asterisk does anything at all to RTP packets
passed from channel to channel if no transcoding is involved? Can
On 04/07/2015 10:48 AM, Johan Wilfer wrote:
Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:
Dear all,
Is anyone has experience making Asterisk server with virtual server
OPEN-VZ (in proxmox 3.4 box) ?
My boss want to build a production server with it, and it will have +/-
300 sip user (concurre
Curious if anyone has any stats on max concurrent calls on different EC2
instance sizes. A client has a proof of concept running on a medium
compute instance now, and we are curious at what point we might
experience issues. All calls are SIP, no transcoding, using SPEEX. I'd
love to hear if
Howdy,
I built an LXC container with an "image" of asterisk 11.18 precompiled
and installed. It runs fine on the dev platform, which is a Dell R320
running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and
untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container
i
. This
is frequently necessary when using in virtual environments.
In cli form: # menuselect/menuselect --disable BUILD_NATIVE
On Wed, Jul 1, 2015 at 1:36 PM, Jeff LaCoursiere <mailto:j...@jeff.net>> wrote:
Howdy,
I built an LXC container with an "image"
Hi,
I have a client that has a 24 channel voice T1 that I have been using
e&m signalling over for a number of years. The local telco finally got
an ISDN switch and wants to move them to PRI. I didn't see this as a
big problem - I've done a few others on this particular Caribbean island
wit
[snip]
So what about system.conf would cause lsdahdi to show "T1" instead of
"PRI" in column two? Just trying to head off any additional problems
once they get their patching sorted out.
The issue is probably with the wanpipe configuration and not with
DAHDI or Asterisk. Run the wanpipe_cf
On 09/30/2015 07:47 AM, Tzafrir Cohen wrote:
[snip]
Right. For Sangoma cards, lsdahdi can't tell if the port is E1 or T1 and
thus calls it "PRI". Note that "PRI" here is a poor name that refers to
the port type itself and not to the signalling in it (which don't have
to be ISDN).
Suggestions? Pa
Hi,
I have a need to pass through SIP headers that start with a particular
prefix, without knowing beforehand what the full name of the header
actually is. For example I need to test for any headers on an inbound
channel that start with "FOO_" and then use SIPAddHeader() to add them
to the
Hi,
Our custom application sets some SIP headers that we want passed to the
called party via asterisk in a simple proxy setup. It works fine for
voice calls, but we also use SIP to send outofcall messages. I notice I
can't use SIP_HEADER() to get those custom SIP headers in outofcall
messa
Fail.
On 10/28/2015 04:42 PM, ama...@sevana.fi wrote:
Hi,
Just checking if my emails reach the list.
Thanks,
Amanda
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On 10/28/2015 06:37 PM, Pete Mundy wrote:
Hi Motty,
Isn't the whole point of the nonce in a SIP registration to ensure the
secret doesn't go on the wire in plain-text? Is this not enough, or
are you looking to hide the username too?
(if so, fair 'nuf, just wondering why :)
Pete
Ps, if so t
On 10/29/2015 04:01 PM, Motty wrote:
On 10/29/2015 01:11 PM, Jeff LaCoursiere wrote:
On 10/28/2015 06:37 PM, Pete Mundy wrote:
Hi Motty,
Isn't the whole point of the nonce in a SIP registration to ensure
the secret doesn't go on the wire in plain-text? Is this not enough,
Here is a funny story. We mostly do hotels in the Caribbean, and one of
our first clients (going on ten years now) used the sample "weather.agi"
that used to be shipped with... asterisk@home? Trixbox? Can't even
recall where we originally got it from.
This perl script uses festival to speak
, December 16, 2015 9:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] weather.agi
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent:
That would be the expensive route. The inexpensive route would be to
buy FXS ethernet gateways, like this:
http://www.voipsupply.com/grandstream-gxw4248. You could then get by
with a single reasonably sized asterisk box (probably two setup as HA)
and no need for expensive cards or complex c
Has anyone created any docker images I might be able to use on EC2 for
load testing an asterisk platform? I started an instance this morning
and was about to load sipp and other tools, and then thought surely
someone must have done this already. I'd like to hammer a platform we
have created
On 02/24/2016 04:49 PM, Steve Edwards wrote:
On Fri, 19 Feb 2016, Jeff LaCoursiere wrote:
Has anyone created any docker images I might be able to use on EC2
for load testing an asterisk platform? I started an instance this
morning and was about to load sipp and other tools, and then thought
And punctuation and grammar skills have we too! Our english be VERY good
On 03/31/2016 02:20 AM, ankur verma wrote:
Have you ever heard of Asterisk Development.There are only few companies in
India which are providing this service and "Anticlock Technologies is one of
them.it is dealing in thi
.
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Jeff LaCoursiere
312 962 5250 desk
815 546 6599 cell
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Hi,
I am dealing with a telco that has recently upgraded from a DMS100
switch to a "Metaswitch", and our PRI no longer passes foreign caller ID
information, i.e. if I place an outbound call with specific caller ID
information not associated with the PRI, it gets replaced with the PRI's
primar
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