similar to: Call on hold disconnects...

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).

This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this

Maybe I'm loosing my mind but I've just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. I'm using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone I'm missing? Cheers, Dean -------------- next part -------------- An

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

Hi I am looking for a good wirless headset to use with the Polycom Soundpoint 501 phone. I would greatly appreciate hearing from anyone with good experiences with a specific device. Thanks -- Jim Freeze

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

Hello. I am looking for a simple Asterisk billing solution. I expect about 50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs

I have been getting occasional reports of dropped calls from the users of our asterisk system. Is there anything I can monitor in my logs or in the console to see when a call is dropped? I'd like to see if these drops coincide with network traffic problems. Thanks, Carl

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

Hello, I'm trying to use the package ttda, wich is involved in text analysis, for my own data about answers in a company survey. I've installed it, as well as ispell, but when trying to use an example: > zz <- file("stupid.txt", "w") # build a data file > cat("{comment - stupid data file} \n" , file = zz) > cat("<uci=1>

Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do prefer a SIP phone. All because it's just too difficult to pull a phone cable into the current conference room. Is there any cheaper SIP

Hello, After upgrading from Ubuntu 16.04 to 18.04 printing via SMB-Kerberos no longer works (printing still works in 18.04 when I print via SMB but I don't want to have the password stored in clear text in /usr/lib/cups/backend/smb). In 16.04 I can just type "lpr file.pdf", but when doing this in 18.04 I get "Password for [myuser] on localhost?" and it expects me to type

Dear all, I'm currently running R on my laptop -- a Lenovo Thinkpad X201 (Intel Core i7 CPU, M620, 2.67 Ghz, 8 GB RAM). The problem is that some of my calculations run for several days sometimes even weeks (mainly simulations over a large parameter space). Depending on the external conditions, my laptop sometimes shuts down due to overheating. I'm now thinking about buying a more

I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Has anyone had any success with the mongrel_secure_download gem? I keep getting "connection reset" errors. -- Cheers, Kevin

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk