Hi,All! I've implemented callback to cellphone using asterisk-java and it is running on tomcat for almost a year (I can share the war file and sources) but one big BUT in usability - I hate typing the phone numbers by hand after asterisk calls back - especially those ones I rarely use and do not remember. Thanks to Nokia there are very usefull feature "send touch tones" available in call menu of N95 so the number can be selected from Address book and sent as touch tones. Anyway it still sucks. Ideally I'd like to be able to select the contact and "request the call back" right from the Contact menu and the phone having a data plan can send short IP message (http request, SOAP message) to initiate call back and if you pick up it dials the Contact's number (or sip url for VoIPs sake) So I am planing to dive into Symbian development - and implement the thing ;-)
-- Igor Ostaptchenko Octanix Canada Inc +1-416-628-9314 x2021 [email protected] On Mon, 2 March 2009, Henry L.Coleman wrote: > Yes, using analog lines is very different. With SIP or IAX lines you can > grab the CID and remotely "block" the call, so the * system never has to > answer the call and hang-up. With analog lines you have to go off-hook to > go on-hook, or put another way you have to answer the call before you can > hang-up. Don't forget that the analog telephone was a product of victorian > thinking, in their infinite wisdom they assumed that in order to hang-up > the phone one would have to be on the phone first! :) > > ----------------- > Henry L. Coleman > [VoIP-PBX.ca] > > ================= > > { Dean Yorke} > > > WEll, this is what i have but it doesn't hang up. > > > > I will try again. Also, I am using analogue lines, does that change it? > > > > On 2-Mar-09, at 9:26 AM, Henry L.Coleman wrote: > >> This is all set up via the web GUI in FreePBX. > >> 1. You set up a SIP or IAX Trunk (if you don't have one already) > >> 2. Create a CallBack (can choose to Call a fixed number or the CID > >> of the Call Back number after x secs) > >> 3. Create a DISA (with return of dial tone if you need it) > >> 4. Create an incoming route for the specific DID and CID combo of > >> the trunk and cell phone. > >> 5. Have this incoming call route guided to your callback module > >> 6. Have your CallBack module route the call to your DISA module > >> > >> Takes about 2-3 munuites to set up > >> > >> Call Flow: > >> > >> 416NXXXXXX --->CallBack---> Hang-up--->10sec.delay--- > >> > >> >CallBack(416NXXXXXX)--->answer > >> > >> call--->DISA--Internal Dial tone ---> Dial-out as per outgoing dial > >> plan--->Hang-up (with possible return > >> to new DISA) > >> > >> Using config files you will need to make sure you have the latest > >> DISA and Callback lib functions and that > >> your dial plan will trigger the Callback on a specific DID and CID > >> combo (it's a whole lot more work) > >> > >> Hope this helps H > >> > >> ----------------- > >> Henry L. Coleman > >> [VoIP-PBX.ca] > >> > >> ================= > >> > >> { Dean Yorke} > >> > >>> can you send me some config files. > >>> > >>> I have this kind of working, but it answers the call. > >>> > >>> Thanks > >>> > >>> On 1-Mar-09, at 9:49 PM, Henry L.Coleman wrote: > >>>> I have this up and running on my system PIAF with FreePBX Web GUI > >>>> It's great if you have a cell phone with an unlimited incoming plan. > >>>> You simply call your VoIP line number (DID) from your cell phone and > >>>> route > >>>> the call to do a "Call-Back" based on the CID (of your cell phone) > >>>> The * system doesn't answer the call so you won't get charged for > >>>> the call. > >>>> After 10 secs it calls me back with DISA dial tone. I can then dial > >>>> any number > >>>> I like at the VoIP call rate. The lastest version of FreePBX can > >>>> even return dial tone to make another > >>>> call without going through dialing a call back again. > >>>> > >>>> Trust me ..this one is very cool! > >>>> ================================= > >>>> Henry L.Coleman [www.VoIP-PBX.ca] > >>>> Tel: 647-723-5160 Ext.203 > >>>> ================================= > >>>> > >>>> > >>>> { Bill Sandiford} > >>>> > >>>>> Bruce: > >>>>> > >>>>> I certainly can't speak to what Comwave is using specifically, but > >>>>> there are many platforms out there that > >>>>> can offer this type of service (including Asterisk). > >>>>> > >>>>> Bill > >>>>> > >>>>> -----Original Message----- > >>>>> From: Bruce N [mailto:[email protected]] > >>>>> Sent: Sunday, March 01, 2009 6:40 PM > >>>>> To: asterisk Mailing > >>>>> Subject: [on-asterisk] Comwave Callback and Similar > >>>>> > >>>>> > >>>>> Hello Everyone, > >>>>> > >>>>> In the past few months Comwave has been advertising about a sort of > >>>>> CALLBACK/DISA combination for > >>>>> $20/month. Can anyone speculate as to what sort of equipment/ > >>>>> solution they use to achieve their goal? Even > >>>>> if they sign up 100k clients that is still a lot of calls knowing > >>>>> every call is actually two calls going > >>>>> through their system and then bridged. Do they use an Asterisk SER > >>>>> farm? Do they use some propriety > >>>>> equipment like VPS, MERA, etc...? > >>>>> > >>>>> Listed: > >>>>> http://www.comwave.net/mobile/ > >>>>> > >>>>> Thanks, > >>>>> Bruce > >>>>> --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
