Grandstream HT386 also has that feature. Into the configuration you can find a field called 'Audial Off-hook', there you can set any extension so the ATA will dial as soon as you pick up the handset.
On 8/6/07, James FitzGibbon <[EMAIL PROTECTED]> wrote: > On 8/3/07, Michael Munger <[EMAIL PROTECTED]> wrote: > > > > > > > > > > Is there a way to setup an IAX bat phone (immediate=yes) or is this a > privilege only reserved for ZAP channels? > > As I understand it, this would have to be supported by your specific > hard/soft phone. > > It's the same with SIP - taking a handset off-hook doesn't cause any traffic > to go to Asterisk. The first packet from the user agent is sent when the > phone tries to dial something. Depending on the user agent, this could be > as soon as someone presses a single key (so-called "early dial" with SIP 484 > responses), or more typically when an entire number has been dialed and a > timeout has occurred or send button has been pressed. Zap FXS ports can > tell when a handset has gone off-hook and take some action based on that due > to the change in electrical impedance. > > Some soft-phones support bat-phone operation, though you have to hunt > through the docs to get it to work. My Linksys SPA942 desk phone has a dial > plan syntax that allows this: > > (<:XXXX>S0) > > Which means "prefix whatever I type with XXXX and match an empty string, > dialing as soon as you have a match", which causes the phone to calll XXXX > as soon as I take it off hook. But it's obviously device-specific, and has > nothing to do with SIP or IAX or Asterisk for that matter. When the call > arrives at my server, it doesn't look any different than a call to XXXX from > a phone with a more traditional dialplan. > > -- > j. > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users