search for: dialpad

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 before... thanks

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 SIP and I have to live with? Thanks! Craig

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 t...

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 configuring'em also in sip.conf). I think that's a problem of the ISDN line...

...e 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. I think that the inband-dtmf Grandstream issue may cause this problem. Going in the details: I call the 803121 (Telecom Italia toll-free number for business customers). I hear indefinitely the ring. They have an IVR. I call the 800281111 (a toll free number of a popular radio station in italy...

...the restaurant when I get the whole place switched to VOIP, but for now I need something only in the two kitchens. I like the idea of a regular phone with a kick'n speakerphone, but I'm open to alternatives. I say 'regular phone' with unease, but I mean something with a normal dialpad, extra buttons for different functions, handset and speakerphone. I've been considering cisco and polycom. Specifically I've been thinking about the Cisco 7940g or something like it. Also I've been considering the Cisco 7920 in a holster w/ wired headset. I'm welcome to any r...

...c per-campaign + allowed for webpage forms launched from GUI with call info + allowed for different web form web pages per-campaign + allowed agents using VICIDIAL client to send calls with data to a closer + new call-closer functionality is web-based for flexibility + added new DTMF dialpad for sending tones quickly + added code from Paul Concepcion to allow admin pages to work with PHP globals off + added common database connection file for each set of PHP admin pages Let me know what you think. Thanks, MATT---

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 trying to connect. LINPHONE -- Linphone does not work with ALSA, nor wit...

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 in the conference room working offhook. 2. Dialler originates calls from the meetme confer...

...Specification SIP: IETF RFC 3261 Codec: G.711, G.729a/b, G.723 Acoustic echo cancellation Dynamic jitter buffer Voice activity detection Stun-based NAT traversal Input Methods Handwriting Recognition > English > Chinese > Numeric characters Soft Keypads > Qwerty > Standard phone dialpad > Symbol Power Management Features Standby time >100 Hours (GSM on, WLAN on) > 200 Hours (GSM on, WLAN off) Talk time > VoIP Over WiFi: 3.3 Hours > GSM: 7.8 Hours MP3 play time > 5.8 Hours (GSM on, WLAN on) > 6.2 Hours (GSM on, WLAN off) Fixed Mobile Convergence Features Si...

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

...et> Organization: Edvina AB To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News) Reply-To: asterisk-users@lists.digium.com WipeOut wrote: > Anthony Minessale wrote: > >> I found this information on how to make XP have a dialpad in Windows >> Messenger >> which was awesome news > Some more crushing news is if you upgrade MSN messenger past ver 4.x it > no longer uses SIP.. (so I have been told).. MSN messenger does not use SIP. Windows messenger (another product) use SIP. I still can't get Windows...

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

...da works in that if I lift the receiver I hear dialtone and if I call that extension, although the phone does not ring, it does answer when I lift the receiver and can have a conversation with the person the other end. The problem seems to be that any of the phones functions/keys including the dialpad (I cannot use it to dial) do not work. Have I got any chance of getting these to work? Maybe I am just better off throwing them away and re-cabling the building for SIP phones? Thanks, Sam -------------- Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Grea...

...ks right, the lights light up right, and when we press one of the associated buttons that extension is dialed. On the problem phone, the lights light up right but when we press one of the associated buttons, I get a "Forbidden <extension>" error. If I dial a number manually on the dialpad, all works fine. The other thing that I noticed is that the working phone has a black phone icon next to the extension number (identity), which means that the account is registered and the line is active for outgoing calls. On the other phone, the icon is a "white" phone which means tha...

...lbc allow=ulaw allow=alaw allow=gsm context=default canreinvite=no nat=no reinvite=no dtmfmode=info tos=0xB8 [tracey] type=friend disallow=all allow=ulaw allow=alaw dtmfmode=info host=dynamic username=tracey mailbox= context=internal callerid="tracey" <868> <aastra.cfg> live dialpad: 1 suppress dtmf playback: 1 time server1: pro5.pbzinc.loc sip dial plan: "X+#|XX+*" sip proxy ip: phone.pbzinc.loc sip proxy port: 5060 sip dtmf method: 1 sip out-of-band dtmf: 0 sip line1 auth name: tracey sip line1 user name: 868 sip line1 display name: "Tracey" sip line1 scr...

...as a trunk/FXO. However, i want to use the phone as an FXS. Before ending up in trying something that was never foreseen and perhaps even impossible, i was hoping that i could use the BB as an oridinary "audio device" and still use the keys on the phone for starting/ending calls, and the dialpad for selecting phone numbers. And having the connections go (via BT) through asterisk instead of GSM. Is this possible at all, or am i embarking on a "mission impossible" ;-) Hans

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