similar to: Vocera Badges

Hi, has someone ever got their hands on the Comm Badges from Vocera ( http://www.vocera.com/ ) and knows if they use anything standard and could work with asterisk, or does someone know an alternative to their really small, light devices? Regards, Andreas _________________________________________________________________ -------------- next part -------------- An HTML attachment

Hi there, yesterday I came across the "Vocera Communication Badge" and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp ** Wireless Specifications

I know this has been covered more times than to mention and this is where I got most of my info from... But I am having issues with this. I can't seem to get the phone to register with *. This is being tested on a internal network right now. Here is the setup - sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

We have a test server that runs a single PIII 500MHz(256MB RAM) under Slackware, and we can get 12 SIP -> Zap calls running on it just fine. Over that and we have seen intermittant errors like call quality and very high load spikes. You should be able to get at least 24 SIP -> Zap on that setup. Post on the list when you do max it out. It's always good to see capacity specs. MATT---

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 AAAA CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? ----- Andrew Thompson http://aktzero.com/

I just booted Xen unstable (2.6.16 kernel) and Domain0 is running fine, all on Debian 3.1 Sarge unstable on amd64. I can''t do much else: Error: Error connecting to xend: Connection refused. Is xend running? What''s wrong and how do I fix it? -- Neal Probert (nealzilla@probestar.com) aka Niall Lundkvist

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language,

Hi, Anyone used this service, any comments on reliability/support? Thanks John

Can someone tell me what this setting is supposed to be? peer->nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? ----- Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

I saw a few people mentioning they were running or trying to run asterisk on Xen. Last night I checked out v1-0, compiled, ran "make install; make samples", then started asterisk with "asterisk -vvvvc". Several modules refused to load giving this error: [chan_sip.so]May 22 13:48:41 WARNING[4308]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so:

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

Hi all, Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved. For the interested people, you can download the new DLL (just the IAX2 version) from the following location: http://www.laser.com/dante/diax/wiax2.zip Replace the wiax2.dll file in the app directory with the new one and this is all. Please test it and send me your feedback. I intend to release a new DIAX version this

Good <fill in local time of day> I'm looking for a piece of hardware that we can place in two offices that have decent bandwidth, but are in two different US states. There are phone systems on both sides, that have extra CO analog line ports that I'd like to connect through. One side has an IVR, the other side does not use one during office hours. The best configuration would

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd