similar to: Dial by name across 2 systems

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (<100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything >300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct

We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same

Searched the wiki and googled extensively to no avail. How can the Directory App be used by a VoiceMailMain user to forward a voice mail to another voicemail user and do it by a directory 'dial by name' lookup? Ex. A voicemail system with 1000+ subscribers. One user wants to forward another user but has not idea what their extension or voicemail box number is.

I took a little foray into pricing out IP Phones for my home pbx yesterday. $75-$750 seems to be quite a range, so I took a closer look. Cisco, for example, has different models such as the 7940 and 7960 which seem to only differ in the software. And buying a Cisco 7920 should cost you $500 w/o a license, and $200 more with a license? It seems to me that these prices are substantially

Hi, > I've never run against a commercial PBX that didn't need > maintenance. Acknowledged. > VM hard > drives fail, > ... > Asterisk is > every bit as stable > as the old-gen KSUs and PBXs. There are big differences. As I know of no other PBX that uses 'consumer' hardware, asterisk is also struggling with problems in the underlying Hardware. And

Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread:

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

IPManager version 1.6 has just been released. Below is a list of some of the features you will get on your Asterisk server using IPManager to generate your configuration files. Download: http://ipsoftware.thorben.dk <http://ipsoftware.thorben.dk/> PBX Features The following features will be available to users of the PBX if you are using IPManager to configure your PBX. *

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

There has been discussions about the voicemail menus and some of us would like to see an overall plan for the voicemail menus. There are 3 primary ways of arranging the menus. First is a tree structure, second is a random access structure and the third would be a hybrid of the two. (Comedian mail is currently a hybrid.) As was pointed out by Brad Bergman, the ideal would be to have it

this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: "Generic PBXs will not do for our broadcast application ? they just don?t have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch groups of lines from studio to studio. There is also no way to connect computers for

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

I am currently testing an asterisk server with some cisco 7960 phones. I have been having problems with phones rebootin using 6.3 firmware in the asterisk voicemail menus. The phones reboot after a dozen or so random button presses while in the voicemail menus. To try and fix this, I upgraded to sip 7.1 only to find that now the phones reboot even if i'm trying to press a button to

Hi, I''m trying to write a sidebar menu with dynamic menus, and to do so I would like to define a list of menus to display within my sidebar component controller code and pass that list to the component display view, like so: # menu_controller.rb class Sidebar::MenuController < ActionController::Base uses_component_template_root @menus = %w{admin user help} def display

I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the "pre-packaged" commercial PBXs to a more "pure" asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain