similar to: setup multiple phones for 1 extension

I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I

Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas

I've read through the Wiki, I know its hard to nail down hardware requirements because it really depends on what you are going to do. I'm very new to Asterisk, haven't even read my Asterisk for Dummies book yet (its in the mail). Can you tell me if this sounds sane? We are planning on using a Dell 933Mhz dual CPU server, with 1GB of ram for our Trixbox setup. We will have 7-10

Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The

Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone's UI may be able to indicate which account is ringing without the need for BLFs. Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line

I have a share created on my linux server. I can connect to it with the "administrator" user just fine. I need to be able to create files and directories with the administrator user, even in directories that are not owned by administrator. administrator is in the users group. Everything under the share is in the users group, and the group has write permission. So why can't

I have added CID name prefix on inbound route. and it works fine :) now I can simply forward five incoming routes to one extension. and as far as I guess, if I add CID name prefix for every number. it should work :) thanks alot :) On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.network at gmail.com> wrote: > hello Pete Mundy, > > thanks alot for your idea and reply. but

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

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Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has "soft keys" (yes, I know they're all soft - but you know what I mean; programmable buttons whose

Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone).

Here's an interesting issue we're facing... We would like users to be able to use softphones from home/work and to use their same extensions they do at work. The first step of getting the phones to log in as their same extensions as work is easy and works. However, on the database side, once the client closes, the sip table is cleared of the ip to the phone. This means that no calls are

Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the

http://www.wunderground.com/weather/api/ -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of dk at donkelly.biz Sent: Wednesday, December 16, 2015 9:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] weather.agi -----Original Message----- From: