Re: [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
On 16.07.2012 04:17, Michelle Konzack wrote: Hello Armin, Am 2012-07-13 15:17:33, hacktest Du folgendes herunter: On 29.06.2012 09:00, Michelle Konzack wrote: I do not know, because I have not bougth it yet. It is only named: Eicon DIVA Server 4BRI-8M 2.0 800-665-02 version 2.0 If it is really a rev2 (2.0), then the NT mode should work. Thanks. I now have bought it. Will give it a try. Question: Must I set the whole card into NT mode or can I do this per ISDN Port? Each port can be configured individialy. I have the datasheets of the Cologne Chips here and the 2port and 4port chips can set individualy... So, I assume, it is a question of the ISDN card implementer, whether it allow to set individual ports to TE/NT mode I have the Development Kit here with some samples, but failed to build a small arme based GSM/HSPA + ISDN/Analog router... Currently I use the Vodafone Easybox 803A which support generaly what I need, except that 1) it support only one USB GSM/HSPA Stick 2) I can not access the SMS storage 3) support only Analog and ISDN Telephone and Telephone-Systems and the (most important part 4) I can not use VoIP telphones So I would like to build a GSM Router which support the above stuff and of course can run a Linux-Distribution of choice using Asterisk! To start a new Open Hardware Project I search for contributors like an Electronic Engineer and Software Developers (we need special drivers for the stuff) and I prefer to use: 1) TI OMAP 300-700 MHz ARM Microcontroller 2) =3D 128 MByte of SDRAM 3) =3D 512 MByte of NAND 4) Cologne Chip XHFC-2S4U or XHFC-4U (could be a module) 5) SiLabs Quad ProSLIC 6) Cypress 4 Port Tetra Hub 7) 8port or 12port N-Way Switch supporting PoE 8) SDHC or SDXC Card interface maybe with 9) 4 TFT Display (480x272 pixel) for status 10) Integrated 5Ah LiPoly cell (10Ah option) for Backup since the router should work some hours even if there is an electricity cut. Maybe this sounds like a Dream-Box, but is entirely technicaly possibel! And of course, I have already aquired most chips in the last 2 years... Note: I am using CadSoft Eagle under Debian to desingn PCBs and a mailinglist is already available even if unused... If you are interested in Open Hardware Development feel free to contact me, because it is time to change something. Oh, forgotten one thing: I was even thinking to use a Marvel Discovery MV78100 (800/1000/1200MHz) because it support: 1) SO-DIMM Modules 2) huge NAND Flash 3) SD-Card 4) 2 PCIe 4x ports which can be each splited into 4 ports of 1x 5) 2 SATA Drives 6) 1 Giga Ethernet Sounds very interesting! Armin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
Dears;
First, I would like to declare that I used sip_custom.conf and
extensions_custom.conf and I discovered that the calls are not shown in the
CDR.
Until now I did not check if I can browse the voicemails using the FreePBX if I
used the extensions_custom.com file.
Now, if we are talking about using the addon modules (official or not
official), then it means still I have to stick on using the FreePBX GUI which
generates complicated script.
As I see, if I need to have the CDR and the voicemail functionalities that can
be browsed via the GUI, then I have to use the FreePBX or the third party
modules and can not write manual in my hand as normally we do in the native
Asterisk.
Thanks for the input for all the friends who shared with me and gave me the
good information that also helped.
Regards
Bilal
See
Route-Permissions module,
It lets you restrict certain phones/extensions to follow a
dial-plan
pattern and dial out to the defined trunk etc meanwhile not
breaking any
other functionality or features of FPBX- though you can
restrict the
features from this too.
http://www.freepbx.org/support/documentation/howtos/how-to-give-a-particular-extension-different-or-restricted-trunk-access
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/outbound-route-permission
http://mirror.freepbx.org/modules/release/contributed_modules/
OR
Custom Context
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/customcontexts
See w/e fits your requirements. What I suggest suits your
need is the
Route-permission module. Though it'll be bit complicated but
worth giving a
try.
Regards,
Sammy
On Thu, Jul 12, 2012 at 4:01 AM, Warren Selby wcse...@selbytech.com
wrote:
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.comwrote:
Fine, did you read the question well and understand
about what I am
asking?
Perhaps I did not understand what you were
asking. I thought you were
wanting to do something custom per extension (in the
case of my example,
the something custom was control outbound call access
to either local
only or local and long distance, etc. You can
figure out you're own
something custom), but still have all the calls have
all the standard
FreePBX features that you only get when using the
[from-internal] context.
In my example, the extensions are in the 2XXX range,
and they would either
have a context of [custom-local-only] or
[custom-long-distance], depending
on what you wanted to allow that extension to dial.
To break down my example:
[custom-local-only] -- The name of our custom
context. It could be
anything you want, as long as it's in square brackets
exten = _281NXX,1,Verbose(Outbound call from
local-only context) --
This step is purely informational, it has no bearing on
CDRs or anything
else...it's just a useful step for debugging. I
tend to do this for
everything, it's the same as some people use the
NoOp() command to have
debugging information in their CLI output.
same = n,Goto(${EXTEN},from-internal,1)
-- This step sends the call to
the [from-internal] context and handles it exactly as
if you weren't using
any custom call controls. In my example, however,
it will only go there if
it meets the criteria of matching the pattern (in other
words, the call
would have to be placed to a number that matches the
_281NXX pattern).
same = n is a shorthand way of writing exten
= _281NXX,n. It was
added in around 1.6 I think, I'm not entirely sure.
exten = _2XXX,1,Verbose(Internal
extension-to-extension call) -- Again,
this is purely an informational step, useful for
debugging. It can be
skipped or expanded as you see fit, it has no bearing
on CDR records or
anything else, other than CLI output.
same = n,Goto(${EXTEN},from-internal,1)
-- This does the same as the
previous example, however it will only go to the
[from-internal] context if
the pattern that was dialed matches _2XXX. This
is assuming you're using
internal extensions in the range of _2XXX. You
can change this to whatever
works for you.
[custom-long-distance] -- another custom
context, this time it allows
long distance NANPA calling as well as local and
internal calls
exten = _1NXXNXX,1,Verbose(Outbound call from
local and long-distance
context) -- I hope you're seeing the pattern
by now. This is simply a
useful debugging step, with no bearing on anything
else.
same = n,Goto(${EXTEN},from-internal,1)
-- The call passes into the
[from-internal context if it matches the pattern of
_1NXXNXX, a typical
NANPA long distance call.
include = custom-local-only -- include
the local dialing context that
way we don't have to duplicate any code that we've
previously written,
mostly useful for the internal
[asterisk-users] Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10
Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10 ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf #include chan_dahdi_custom.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf system.conf span=1,0,0,ESF,B8ZS bchan=1-23 dchan=24 loadzone=in defaultzone=in -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing auto mixmonitor output file name
Hi, I want to enable call recordings by simply pressing the recording key defined in features.conf. The problem is that I didn't find a way to change the name of the output file, the Dial application has two options for enabling call records, the w and W and x and X. What is the diference between those? I want it this way because We need to record just the important piece of the call, recording the whole call will be a waste of disk space. When I use MixMonitor application, I can change the file name, but how can I do that when the recoding is enabled on-demand? Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10
span=1,0,0,ESF,B8ZS Your timing source is wrong, you should be pulling from your provider. This should be: span=1,1,0,ESF,B8ZS Also, You don't list the error your getting, that would be helpful. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing auto mixmonitor output file name
On Mon, Jul 16, 2012 at 8:48 AM, Antonio Modesto mode...@isimples.com.br wrote: I want to enable call recordings by simply pressing the recording key defined in features.conf. The problem is that I didn't find a way to change the name of the output file, the Dial application has two options for enabling call records, the w and W and x and X. What is the diference between those? I want it this way because We need to record just the important piece of the call, recording the whole call will be a waste of disk space. When I use MixMonitor application, I can change the file name, but how can I do that when the recoding is enabled on-demand? http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Look for [macro-apprecord]. -- thiagoc O povo não deveria temer o governo. O governo é quem deveria temer o povo. V de Vingança -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
Hello Armin Schindler, Am 2012-07-16 09:42:50, hacktest Du folgendes herunter: Each port can be configured individialy. OK Sounds very interesting! Yes, and it is entirely Distribution/Manufacturer independant. I had to stop working in it, because I am hopeless overworked! Since I know the MV78x00 I am dreaming of a Mini-ITX Poard based on this Microcontroller. Could have up to 8 Mini-PCIe slots and a big PCIe 4x for the bigger cards. Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.14.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.14.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- Remove a superfluous and dangerous freeing of an SSL_CTX. (Closes issue ASTERISK-20074. Reported by Trevor Helmsley) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.6.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.6.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.6.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- Remove a superfluous and dangerous freeing of an SSL_CTX. (Closes issue ASTERISK-20074. Reported by Trevor Helmsley) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] any working calling card solution open source
hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I guess my problem is Asterisk-perl I will be glad for a quick response. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any working calling card solution open source
Hello a2billing works fine Regards On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote: hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I guess my problem is Asterisk-perl I will be glad for a quick response. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any working calling card solution open source
thank Carlos, Thanks but too big for a demo interms of setup no demo data. I got the astcc working but still looking for alternative Thanks On Mon, Jul 16, 2012 at 8:32 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello a2billing works fine Regards On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote: hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I guess my problem is Asterisk-perl I will be glad for a quick response. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
