Re: [asterisk-users] STIR/SHAKEN

2021-03-11 Thread Jeff LaCoursiere
a dialplan 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 sure what you

Re: [asterisk-users] STIR/SHAKEN

2021-03-11 Thread Jeff LaCoursiere
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 people showed you a fe

Re: [asterisk-users] STIR/SHAKEN

2021-03-08 Thread Jeff LaCoursiere
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

Re: [asterisk-users] STIR/SHAKEN

2021-03-07 Thread Jeff LaCoursiere
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't even mention call forwarding iss

Re: [asterisk-users] AGI Script Returning 4

2021-01-30 Thread Jeff LaCoursiere
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...@stratustalk.com

Re: [asterisk-users] STIR/SHAKEN

2021-01-28 Thread Jeff LaCoursiere
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: *j...@str

[asterisk-users] STIR/SHAKEN

2021-01-25 Thread Jeff LaCoursiere
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

[asterisk-users] Fwd: Your message to asterisk-users awaits moderator approval

2021-01-25 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

2021-01-25 Thread Jeff LaCoursiere
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, Jan 24, 2021 at 6:50 PM

Re: [asterisk-users] Asterisk registrations - state?

2020-12-15 Thread Jeff LaCoursiere
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:*One Freedom Square 1

Re: [asterisk-users] which linux for asterisk?

2020-12-09 Thread Jeff LaCoursiere
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 stable platform

Re: [asterisk-users] which linux for asterisk?

2020-12-09 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

2020-10-30 Thread Jeff LaCoursiere
re 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] type = f

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

2020-10-30 Thread Jeff LaCoursiere
re 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] type = f

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

2020-10-29 Thread Jeff LaCoursiere
re 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 wo

Re: [asterisk-users] Stir Shaken

2020-07-13 Thread Jeff LaCoursiere
g 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-digital.com --

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-23 Thread Jeff LaCoursiere
? 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:*https://www.stratusta

Re: [asterisk-users] Voice "broken" during calls

2020-06-22 Thread Jeff LaCoursiere
it 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 possible. Please check your Fritbox, I can't reach it"... ;) Bye Luca Bertonc

Re: [asterisk-users] Voice "broken" during calls

2020-06-16 Thread Jeff LaCoursiere
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?

Re: [asterisk-users] Voice "broken" during calls

2020-06-15 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Voice "broken" during calls

2020-06-15 Thread Jeff LaCoursiere
, 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://www.stratustalk.com* A

Re: [asterisk-users] CDR mysql: timeout when remote database unavailable

2020-06-07 Thread Jeff LaCoursiere
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,

Re: [asterisk-users] CLI color prompt

2020-06-01 Thread Jeff LaCoursiere
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* <mailt

Re: [asterisk-users] CLI color prompt

2020-05-31 Thread Jeff LaCoursiere
(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.com> Website:

Re: [asterisk-users] CLI color prompt

2020-05-31 Thread Jeff LaCoursiere
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: *j

Re: [asterisk-users] Stir-Shaken for asterisk

2020-05-27 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Stir-Shaken for asterisk

2020-05-27 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Block Spam Calls

2019-12-13 Thread Jeff LaCoursiere
, *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*

[asterisk-users] Delayed RTP start

2019-07-24 Thread Jeff LaCoursiere
' 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 Stratu

Re: [asterisk-users] Hacking

2019-06-18 Thread Jeff LaCoursiere
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

Re: [asterisk-users] DUNDI with minimal features

2019-03-27 Thread Jeff LaCoursiere
/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 815 546 6599 cell j...@stratustalk.com

Re: [asterisk-users] Digium G100

2019-02-15 Thread Jeff LaCoursiere
twork 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] Digium G100 Hi, W

[asterisk-users] Digium G100

2019-02-14 Thread Jeff LaCoursiere
rom 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.api-d

[asterisk-users] Digium T1 gateway caller ID issues

2019-02-13 Thread Jeff LaCoursiere
rom 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.api-d

[asterisk-users] Xorcom PRI

2018-11-12 Thread Jeff LaCoursiere
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

Re: [asterisk-users] multi step auth?

2018-05-08 Thread Jeff LaCoursiere
/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 <j...@stratustalk.com <mailto:j...@stratustalk.com>> wrote: I *am* doing that,

Re: [asterisk-users] multi step auth?

2018-05-08 Thread Jeff LaCoursiere
ofcourse for each customer you will need to provide his own did. On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere <j...@stratustalk.com <mailto:j...@stratustalk.com>> wrote: Hi, We have been using Voxbone for some time for origination, and they now offer E911 services. 

[asterisk-users] multi step auth?

2018-05-08 Thread Jeff LaCoursiere
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

Re: [asterisk-users] ​ PJSIP and Non Media Proxy

2017-11-06 Thread Jeff LaCoursiere
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

[asterisk-users] Openfire and asterisk

2017-10-09 Thread Jeff LaCoursiere
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

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Jeff LaCoursiere
On 05/31/2017 04:13 PM, Steve Edwards wrote: On Wed, 31 May 2017, Barry Flanagan wrote: sngrep Isn't sngrep a great tool? Since discovering it my use of tcpdump/wireshark has cratered. Being able to compare an INVITE that worked with one that didn't (with color highlighting) rocks. On

Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

2017-04-29 Thread Jeff LaCoursiere
2017 at 17:11, Jeff LaCoursiere <j...@jeff.net> wrote: On 04/29/2017 10:57 AM, Jonathan H wrote: On 29 April 2017 at 16:47, Tech Support <aster...@voipbusiness.us> wrote: I’m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it

Re: [asterisk-users] configure AudioCodes MP-112 with Asterisk.

2017-04-29 Thread Jeff LaCoursiere
On 04/29/2017 11:12 AM, the...@sys-concept.com wrote: I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial

Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

2017-04-29 Thread Jeff LaCoursiere
On 04/29/2017 10:57 AM, Jonathan H wrote: On 29 April 2017 at 16:47, Tech Support wrote: I’m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it, I’m getting hundreds and hundreds of errors. Here is a sample of

Re: [asterisk-users] AGI Exec Voicemail

2017-04-12 Thread Jeff LaCoursiere
- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, April 12, 2017 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [asterisk-users] AG

[asterisk-users] AGI Exec Voicemail

2017-04-12 Thread Jeff LaCoursiere
Hi, I have a voicemail broadcast AGI that has been running fine for years - it collects extensions and then EXECs the Voicemail app, like this: EXEC Voicemail \"%s\" (%s is the extension list like AAA etc) This works fine, but after leaving the message and pressing "#", I just get "Thank

[asterisk-users] E-911

2017-03-02 Thread Jeff LaCoursiere
, -- Jeff LaCoursiere 312 962 5250 desk 815 546 6599 cell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start

[asterisk-users] multiple outbound invites

2017-02-22 Thread Jeff LaCoursiere
Hello, I have two upstream providers we use for US termination. The dialplan sends calls out the "primary" and if that fails for specific reasons, it sends the same call out the "secondary". This has worked well for us when we are lazy about keeping balances up, for example. Starting a

[asterisk-users] inbound T38 to email

2016-11-30 Thread Jeff LaCoursiere
. I vaguely remembered a 't38modem' project on sourceforge and integration with hylafax, and started looking at that today, but t38modem hasn't been touched since 2009. Is there any new modern way to take t38 from a (SIP) DID provider and route to email? Thanks for any insight :) -- Jeff

[asterisk-users] new inbound DID provider... no auth?

2016-11-30 Thread Jeff LaCoursiere
potential addresses without authentication info? Cheers, -- Jeff LaCoursiere 312 962 5250 desk 815 546 6599 cell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk commu

Re: [asterisk-users] Metaswitch caller ID passing

2016-08-06 Thread Jeff LaCoursiere
isk instances to send foreign caller ID information and it was accepted. Cheers, j On 08/05/2016 09:05 AM, Jeff LaCoursiere wrote: 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 infor

[asterisk-users] Metaswitch caller ID passing

2016-08-05 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Asterisk and Yealink T21P E2

2016-07-14 Thread Jeff LaCoursiere
terres Sure. Tons of them. -- Jeff LaCoursiere 312 962 5250 desk 815 546 6599 cell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Asterisk Development Company in India

2016-03-31 Thread Jeff LaCoursiere
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

Re: [asterisk-users] load test docker images?

2016-02-24 Thread Jeff LaCoursiere
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

[asterisk-users] load test docker images?

2016-02-19 Thread Jeff LaCoursiere
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

Re: [asterisk-users] 1000 analogue lines with asterisk

2016-02-17 Thread Jeff LaCoursiere
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

Re: [asterisk-users] weather.agi

2015-12-16 Thread Jeff LaCoursiere
, 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: Wednesday

[asterisk-users] weather.agi

2015-12-16 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Asterisk encrypted authentication for clients

2015-10-30 Thread Jeff LaCoursiere
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, or are you

Re: [asterisk-users] Asterisk encrypted authentication for clients

2015-10-29 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Test

2015-10-28 Thread Jeff LaCoursiere
Fail. On 10/28/2015 04:42 PM, ama...@sevana.fi wrote: Hi, Just checking if my emails reach the list. Thanks, Amanda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] SIP headers in outofcall messages

2015-10-26 Thread Jeff LaCoursiere
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

[asterisk-users] partially match a SIP header

2015-10-23 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Xorcom T1 to PRI

2015-09-30 Thread Jeff LaCoursiere
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?

[asterisk-users] Xorcom T1 to PRI

2015-09-24 Thread Jeff LaCoursiere
Hi, I have a client that has a 24 channel voice T1 that I have been using e 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

Re: [asterisk-users] Xorcom T1 to PRI

2015-09-24 Thread Jeff LaCoursiere
[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

[asterisk-users] Dell portability

2015-07-01 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Dell portability

2015-07-01 Thread Jeff LaCoursiere
. 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 j...@jeff.net mailto:j...@jeff.net wrote: Howdy, I built an LXC container with an image of asterisk 11.18

[asterisk-users] use of EC2

2015-04-08 Thread Jeff LaCoursiere
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

Re: [asterisk-users] OpenVZ with asterisk 13

2015-04-07 Thread Jeff LaCoursiere
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

Re: [asterisk-users] RTP handling

2015-03-24 Thread Jeff LaCoursiere
On 03/24/2015 04:28 PM, Richard Mudgett wrote: On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net 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

[asterisk-users] RTP handling

2015-03-24 Thread Jeff LaCoursiere
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

Re: [asterisk-users] PRI Callerid Passthrough

2015-03-18 Thread Jeff LaCoursiere
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

Re: [asterisk-users] PRI Callerid Passthrough

2015-03-18 Thread Jeff LaCoursiere
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?

[asterisk-users] DND on a Polycom IP450

2015-03-09 Thread Jeff LaCoursiere
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

Re: [asterisk-users] AWS/EC2 server selection

2015-03-08 Thread Jeff LaCoursiere
,* 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/EC2 servers. Requirement is to run

Re: [asterisk-users] AWS/EC2 server selection

2015-03-08 Thread Jeff LaCoursiere
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 j...@jeff.net mailto:j...@jeff.net wrote: Amazon instances

Re: [asterisk-users] AWS/EC2 server selection

2015-03-06 Thread Jeff LaCoursiere
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

Re: [asterisk-users] [OT] switches

2015-02-25 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Astricom 2014 presentations

2014-10-29 Thread Jeff LaCoursiere
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 --

Re: [asterisk-users] Playback/background audio from MySQL BLOB

2014-09-24 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Playback/background audio from MySQL BLOB

2014-09-23 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

2014-09-02 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

2014-09-02 Thread Jeff LaCoursiere
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,

Re: [asterisk-users] OT: Question on Caller ID (Spoofing calls with Asterisk)

2014-08-26 Thread Jeff LaCoursiere
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

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread Jeff LaCoursiere
On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net 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

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread Jeff LaCoursiere
[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 Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread Jeff LaCoursiere
. --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 text

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread Jeff LaCoursiere
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 j

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread 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

Re: [asterisk-users] PRI timing settings

2014-08-20 Thread Jeff LaCoursiere
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

[asterisk-users] Way to dump PRI settings?

2014-08-19 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Way to dump PRI settings?

2014-08-19 Thread Jeff LaCoursiere
-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 settings

[asterisk-users] PRI timing settings

2014-08-19 Thread Jeff LaCoursiere
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 show

Re: [asterisk-users] Numbers hackers call

2014-03-26 Thread Jeff LaCoursiere
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),

Re: [asterisk-users] IAXModem or T38Modem?

2014-03-25 Thread Jeff LaCoursiere
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

Re: [asterisk-users] IAXModem or T38Modem?

2014-03-25 Thread Jeff LaCoursiere
On 03/25/2014 10:13 AM, Steven Howes wrote: On 25 Mar 2014, at 15:00, Jeff LaCoursiere j...@jeff.net 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

Re: [asterisk-users] stopping unwanted attempts

2014-01-20 Thread Jeff LaCoursiere
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

Re: [asterisk-users] Text to Speech Engine

2014-01-10 Thread Jeff LaCoursiere
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,

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