similar to: CLASS feature syntax

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Source: http://www.nanpa.com/number_resource_info/vsc_assignments.html See also: http://bugs.digium.com/bug_view_page.php?bug_id=0000071 Some (which?) of the codes below are hardcoded into Zap channels only. Is there a European equivalent for this (or ITU / IETF)? Greetings, Philipp VERTICAL SERVICE CODES (US Standard) ASSIGNMENTS *00 - Inward Voice Activated Services (English) *01 -

Hi, I need dialplan to set INCOMING call forwarding during lunch break to my secretary. I want that I can set call forwarding by dialing an extension number to turn it ON or OFF. I am using asterisk 11. Thanks Abdullah Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database with City and State. Actually it's for an Asterisk routing app I'm working on. I see several vendors that want a few bucks to those that want an arm and leg. I expect this is published somewhere by some government agency, but Google hasn't got me to it yet. Jon -------------- next part --------------

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Hi, I am trying to setup some LCR functions on my Asterisk box and have a cheap call provider that uses various different numbers for landlines and cell phone numbers in various countrys.. I am finding it difficult to find the various codes.. eg. UK Landline - +44[12]. UK Cell - +44[7]. SA Landline - +27[1-6]. SA Cell - +27[78]. Please send me your country's dialing rules similar to how I

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

Hello; I've been playing with a nifty Open Source java based report writer called Datavision (datavision.sourceforge.net) and I've managed to write enough logic to calculate phone bills at different rates from the MySQL cdr's. (cdr_addon_mysql) Eventually I want to have sets of rate structures for each user of the system - so I can bill client A at 3 cents a minute and client B at 2

Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

http://bugs.freedesktop.org/show_bug.cgi?id=18336 Summary: nv10 : doesn't detect connected dvi monitor Product: xorg Version: 7.3 Platform: Other OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

As I understand it (and my understanding is obviously incorrect) the switch => statement sells the Asterisk box to resolve (aka lookup) extensions by querying the remote Asterisk server defined in the switch => statement. The switch => statement is used to centralize dialplans. I've not used the switch => statement yet, I'm just trying to understand the ramifications of using

I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

Does anyone have a mirror of this running? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050715/5fe78cd2/attachment.htm