Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

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

Re: [asterisk-users] Get channel variables via ARI/AMI

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

Re: [asterisk-users] Get channel variables via ARI/AMI

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

Re: [asterisk-users] Get channel variables via ARI/AMI

y 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 815 546

[asterisk-users] PMS integration

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

Re: [asterisk-users] Intro and question

ur 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 -- _ --

Re: [asterisk-users] github - mlan

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, >> >&

[asterisk-users] github - mlan

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. 703 496

Re: [asterisk-users] Run asterisk -rx "command" and get plain text output

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

Re: [asterisk-users] Delay when dialing...

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 and Colocation

Re: [asterisk-users] STIR/SHAKEN

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

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

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

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

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

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

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

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)

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?

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?

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?

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

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

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

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

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...)

? 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

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

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

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

, 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

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

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

(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

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

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

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

, *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

' 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

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

/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

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

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

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

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 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?

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?

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

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

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

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 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.

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

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

- 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

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

, -- 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

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

. 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?

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

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

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

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

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?

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?

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

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

, 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

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

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

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

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

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

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

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

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

[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

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

. 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

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

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

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

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

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

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

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

,* 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

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

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

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

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

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

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

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

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)

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

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

[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

. --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

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

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