search for: softbutton

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard functions seem ok). The Snom 200 I have dials "sip:asterisk@asterisk_IP" which I have an extension...

...390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the default screen when I'm done making a call. Does anyone know how to fix this? --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x211...

Folks, I am in the process of fighting my way through my first foray into ADSI phones. I have a very nice Aastra 480 from Sayson, purchased through Cylogistics. Very nice phone. However, my experience in configuring ADSI has been pure pain so far, and it's not all working yet... 1) The locking mechanism for these ADSI phones is... baroque? Arcane? Perhaps there is a reason why you would

...xt stage:- The colour of the backlight:- I think blue would work best? On/Off Status, I've put some thought into this and think the light should come on in the following circumstances:- Phone Rings Phone handset is picked up Speaker button is pressed Individual Line button is pressed Any softbutton press (4 buttons under display) When light comes on it stays on for 15 seconds if it's a button press or until answered if phone is ringing. Dimming facility:- I can't see an easy way of implementing a dimming facility, any thoughts? That's where I am now....anyone have any thoughts...

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

...one (I'm assuming the others as well), will freeze and can be reproduced if all of the following is true: 1. You enable auto answer in the configurations 2. You call from an Aastra phone to another Aastra phone using the alertinfo that auto answers the phone. 3. You hangup and then try using a softbutton programmed as a speeddial, the phone freezes right then. If I put canreinvite=no in sip.conf, then all is fine. Also even when not using auto answer, the phone will freeze after just a few phone calls if canreinvite=yes. Has anybody else seen this problem? I would realy like to disable reinvites...

..., I'm trying out a Digium D70 phone with Asterisk 11. My voicemail messages are listed in the "visual voicemail" app on the phone, but they do not successfully play back. The correct duration is shown, but the progress bar just jumps back to zero when I press the "Play" softbutton. I can hear my messages fine if I "manually" dial into my voicemail extension. I have "format=wav49" in voicemail.conf. Is that a problem format for the D70? - Mike D70 Current Firmware Version: 1_3_0_2_54153 Asterisk 11.3.0

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

...#39;, '11', and '10' and send them immediately to > asterisk. You can manipulate the digit map on the web > interface, or in the XML files (sip.cfg). > > As a stopgap measure, you can tell your users to just dial > when the phone is on the hook and press the dial softbutton. > This will bypass the digitmap check. > Noah, Terribly sorry...it was late when I wrote that...I meant to say the dialplan.digitmap in the Polycom sip.cfg file, not the Asterisk dialplan. I as having the same issues as you describe - turns out I needed to pay closer attention to the...

> > Jason Brown wrote: > | Anyone have experiece with polycom phones? > | > | I am experiencing a really weird problem. In an office where I have > | the following extensions: > | On the Polycom phones, when I want to dial from extension > 100 to any > | extension 120 or above, or dial out, it dials just fine. If > I want to > | dial from extension 100 to

Hey all...I'm trying to find a phone that will support multiple MWI so that I can have a shared central phone with say 4 users who can see visually that hey have messages waiting. Is there any phone that will do this possibly by re-assigning a soft-button? Can the Polycoms do this since those seem to be the phone of choice these days? Thanks!

...ension) rather than only putting the call on hold. Transfer seems to work fine, but would like to see blind transfer (transfer direct to the remote party, rather than the current behavior where you are bridged to the called party, then have to press transfer again to complete the transfer) Maybe a softbutton for blind transfer, and reserve the hard button for attended transfers? It would be nice if we could change the behavior of the transfer button to do this. Conference works, but I have some trouble dropping the correct party from conversation. Also, when conferenced, I am no longer able to send DT...

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14