similar to: chan_sip and 2 devices under same extension - transferring call endpoint(s)

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

Sounds like a job for TAPI. Google TAPI for Asterisk or Asterisk TSP I've been playing with SIPTAPI and it works pretty well. It's very simple to install and set up. Hope this Helps PC... From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ryan Wagoner Sent: Monday, December 29, 2014 9:52 PM To: Asterisk Users Mailing

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), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr at gmail.com> wrote: > As the handsets have no LCD's to show the dialled number, > I want to give the workforce the ability to dial OUT using the softphone, > (as in, copy/paste the number from the CRM software into softphone then > *immediately* transfer the originated call 'endpoint' to the handset of > the

Hi, please whoever has some expertise in choice of BRI ISDN cards, please restore my faith in community support :) (on private email I can probably explain more than fits for a public forum) Most I'd like to ask is about what to choose, out of what is available... My locality is United Kingdom, lines from British Telecom (BT), but any advice / pointers (I googled around already) are

This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for.... (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the

Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission

Checkout Beronet ISDN cards and Berofix Gateway (appliance or pci card) Personally for my last installation I chose Berofix card which is rock solid and reliable, yet easily configurable. With berofix you don't need telephony drivers on the host system, the isdn card is detected as a NIC and all configuration is done using a web interface. Then you configure FreePBX a new trunk to use the

Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

Anyone got any pointers on where to find USB handsets or headsets that can be used as the audio device on a softphone? Later..

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

I've a problem to communicate from a trunk to a another trunk. My config is : PBX 1 with subscribers 100 - 119 (SIP trunk to asterisk) PBX 2 with subscribers 200 - 219 (SIP trunk to asterisk) Asterisk with Softphone 300 - 319 I can call from Softphone to PBX 1 and from Softphone to PBX 2 I can call from PBX 1 to asterisk and from PBX 2 to asterisk I would like to make call from PBX 1 to PBX

that was my first idea. and how should an other user know which number he should dial? user a: soft phone extension 100 hardware phone extension 101 On 06.02.19 15:25, Mitch Claborn wrote: > You can do this in the dial plan. Register the devices separately and > include both addresses in the Dial() command. > > > Mitch > > On 2/6/19 8:16 AM, basti wrote: >> In

Even better you can set your firefly softphone to auto answer so that you don't even need to be near the pc to answer. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Linn Boyd Sent: Friday, March 04, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi folks, well the easiest soft phone I have found (from this list) is the x-ten. great little piece of software. BUT, on a windows machine I need to plug in a USB device and find the software to drive it which can actually use a normal handset (similar to the s100p) but can be used to actually generate/answer the calls and actually rings when a call is received. Any ideas ?? Gary .