search for: dialpads

Hello: I would like to know if anybody has test various usb phones with an usb hub conected to an asterisk server. The idea is to build a public telephony site with one or 2 analog cards 4 or 8 FXO, least cost routing and billing and usb handsets with dialpad for users, so I really don't need a graphical interface for the soft phone ( the users just dial from dialpad). Any has done this

Hey - I know there's some other people out there that have the 9133i ... has anyone gotten the DTMF tones to work after the far side picks up? I didn't have any problems out of the box with my SPA-841 phones... the aastra has been nicer so far, but I can't seem to get it to dial the touch tones after an auto-answer device picks up on the far side... I googled, to no avail. -Karl

This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of

I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

Greetings, I need a 2 way intercom for separate kitchens to communicate without having to walk back and forth. The speaker has to be loud but clear, not distorted. Sometimes the kitchens can be noisy. It needs to be easy to use. It needs to be easy to clean. It would be nice if it used POE. Eventually I would like the kitchens to be able to dial different parts of the restaurant when I

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

I had this same problem with WINXP WinMESS, (what a name mess) I changed the Distro from Redhat 8.0 to Mandrake 9.1 and bam! It all works!! Does anyone know of a problem with this and RH 8.0???? Are you running Redhat?? I now have Messenger working fine as well as X-ten, Sipps, and some others. I have standardized on Mandrake 9.1 and asterisk seams to have NO problems. REDHat 8.0 proved as

I see there is the DigitTimeout application that sets the maximum time between digits before asterisk will interpet. Is there any way to control the minimum? We are having problems with incoming calls on our X100P where callers try to dial 10, but the 1 gets detected twice and they end up on extension 11. Thanks Mark Farver

Hi all, Has anyone had any experience getting asterisk to work with legacy phones? I have been trying to get some of our exisiting Plessey PKS2 (Masterline) Keystation phones to work with asterisk. They are 4 wire extensions (2 for voice and 2 for data) I have connected a test phone to the asterisk server via a digium 400tdm and it kinda works in that if I lift the receiver I hear dialtone

I have a strange problem with a Snom 360. I have a pair of them and one is acting just fine. The other one has 2 problems, one of which I think is a configuration issues, but I cannot track it down. The main problem with that phone is that the display is wonky. When I last saw it, with the display all the way down and after a power cycle, it was fine. But just before that, the display looked

I'm having problems getting my aastra 480i to register with the asterisk server. I can inititate calls from the phone, but >sip show peers does not show any IP address registered for this phone. I am probably missing something stupidly simple. Anyone have an example config to share or corrections for my configuration? Asterisk 1.2 aastra 480i CT has the 1.4 firmware <sip.conf>

Hi all, Finally i got hold of some bt-dongles that seems p[retty stable, the asus-bt211. After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres and blackberry address) and "mobile show devices" is showing me that the BT-link is up, and remains stable up. Seems good, but it looks like asterisk is seeing the BB as a trunk/FXO. However, i want to use the phone as an FXS.

Hi all, I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It works: I can receive and make calls. But some buttons of USB phone don't work properly. In particular, button *, #, and hangup have wrong key mapping. Someone have tried a USB phone ???? Thamks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it