search for: deskphones

Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005

Hi All we are using Asterisk 1.6.0.9 version.try to use Minivm for voicemail, but having following problem. 1.How any extension let's say 7001 can access his voicemail box from his deskphone, any config or dial plan example is there. What kind of config require in extension.conf,minivm.conf & sip.conf 2. same way for MWI on respective IP Phone what we require to config in

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

Hi, I have some problems with a Queue and state interface. In the queue we have 7 deskphones and 7 cordless. Every user have one deskphone and one cordless phone. I have set up every user with hints, eg: exten => 802,hint,SIP/9144-802&SIP/9144-902 exten => 902,hint,SIP/9144-902&SIP/9144-802 Queue_members: Queue_name Interface...

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

...with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to eachother. Our supplier told us they dont send SDP with the INVITE. Can this be the problem causing dect to dect calls to always use asterisk in the RTP path ? -- Michiel van Baak michiel@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit....

...g if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage scenario in mind, is it worth the extra $800 for the echo cancellation? As a sidenote, I assume these cards are PRI-compatible, correct? Thanks! -erik -- Erik Anderson http://andersonfam.org

...ice starts. FYI: Churches are the _perfect_ example of a distributed business environment where Asterisk shines. What other company do you know of that has such a number of workers who don't have an office in the building? Our church is currently moving - possession June 30 - and we will have deskphones for offices, deskphones at home for key ministry leaders, softphones for minor ministry leaders and/or phantom VM's with email attachments. Don't for get aliases to job functions! People in ministry and volunteer can come/go. Make sure your IVR has name and functions. Somebody will call lo...

I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

I'm looking to use Asterisk to replace my current PBX system. I'm in Australia so I need to use Austel approved equipment. My plan is roughly as follows: - Get a box with a suitable card and install Asterisk - Connect our existing PSTN lines to the Asterisk box - Get suitable softphones/ip phones to connect to the Asterisk system - Route all intra-office calls over our VPN - Route all

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out

Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/1d2cc49a/attachment.htm

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.1...

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

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