How To Make Voip Client Cannot Use Same Username?

Report
Question

Hi All,

I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too.

what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) --> ok then in client 3, i used (1001) same username with client 1. when client 1 is connecting with client 2, my client 3 could make a call to with client…

Asterisk Users 9 months ago 3 Answer

Asterisk VoIP Software 12.0.0-beta2 Now Available!

Report
Question

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 12.0.0. You can immediately download this release at http://downloads.asterisk.org/pub/telephony/asterisk/releases We strongly encourage all interested Asterisk users to participate throughout the testing process. For any issues you might find, please use the issue tracker to report it: https://issues.asterisk.org/jira. We would like you to come to the #asterisk-bugs channel in order to help communicating issues you found. Also, it is also very useful to see successful test reports. You can use the asterisk-dev mailing list for that (http://lists.digium.com). The next major release in the series of our favorite VoIP software will be Asterisk 12,…

Asterisk Announces 1.6 years ago 0 Answer

Moving Calls From One E1 To Another

Report
Question

Hello everyone. I want to migrate an old PBX which uses the Е1-PRI from one Telecom to VOIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is Е1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to connect a PC with 2 ports E1 module between the PBX and the old telecom. One port of the module will be connected to the telecom's wire and the…

Asterisk Users 2.1 years ago 1 Answer

Asterisk Sizing For Play And Dtmf Detection

Report
Question

Dear all

i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call)

The IVR service is based only on DTMF tones so the features required is

- play feature - dtmf detection

Asterisk will receive calls via VOIP (SIP with g711 codec)

The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections.

How many simultaneous call can i handle per server? each server will have:

4 core 3.0 Ghz 4 GB of RAM

I need an…

Asterisk Users 2.4 years ago 3 Answer

Playing Music Through VoIP Handsets While On Hook

Report
Question

This is something I've seen with some key systems and PBXs. When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it. Never heard of it being done with VoIP, but figured I'd ask if anyone else has. I don't see any way to do this on any phones I've looked at.

Asterisk Users 2.5 years ago 5 Answer