If you're using a php i'd take a look at phpagi - there are others around for various different languages too. our agents use twinkle with auto-answer, the only reason they need to look at twinkle is if they need to perform a transfer (that too will soon be done from the web browser), you can do pretty much anything with the asterisk manager (originate the call and hangup the call and a load of other useful stuff)
Cheers 2009/3/10 Kurian Thayil <[email protected]> > Hi Steve, > > That worked beautifully. Thank you so much. But one question though. > Imagine if I keep a Hangup Button in the interface and it should terminate > the call. Will that be possible? This scenario happens when the user gets > connected to an invalid phone number where the user have to manually > disconnect. I don't plan to confuse the user by asking them to use eyebeam > to disconnect the call. If it could be integrated to the web interface they > just have to stick on to that alone. Is there any way? > > Regards, > > Kurian Mathew Thayil. > > On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro < > [email protected]> wrote: > >> >> >> On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil <[email protected] >> > wrote: >> >>> Hi All, >>> >>> Is there a way that I can include call dialing functionality in a >>> webinterface. I have EyeBeam configured with a SIP user say >>> 8440. Will I be able to design an inteface which agent can choose a >>> number and the Dial without punching in the number in >>> Eyebeam. >>> I tried using the .call file. ie The agent can choose which number to >>> dial from a web interface. Then, a .call file is >>> created with the following informations. >>> >>> Channel: Zap/g2/9444204943 >>> Context: inbound_support >>> Extension: 8440 >>> Priority: 0 >>> >>> Now, in the extensions.conf file, I mentioned the following under >>> inbound_support context. >>> >>> [inbound_support] >>> exten =>8440,1,Dial(SIP/8440,55,tTo) >>> exten =>8440,2,Answer >>> exten =>8440,3,Hangup >>> >>> But, here the call gets connected only when the receiver end receives the >>> call. When the receiver end picks up the phone, >>> SIP/8440 rings. >>> >>> Is there any other way to implement this. I am not ready to use Vicidial >>> (AstGUIClient) because the interface to be designed >>> is too custom and the agent should have the list of numbers in front of >>> them while they dial which cannot be done using >>> Vicicial. >>> >>> Regards, >>> >>> Kurian Mathew Thayil. >>> >> >> The following will ring the internal support personnel (8440) first, after >> answered, it will then dial the customer (14109850123) (Are you in >> Maryland?) >> >> Turn on auto-answer and it should be seamless. >> >> >> Stolen from Wiki: >> >> To create a call to 14109850123 on a SIP phones called bt101, here's the >> file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of >> course must be accessible and deletable by asterisk GNU/Linux user): >> >> Channel: SIP/8440 >> MaxRetries: 1 >> RetryTime: 60 >> WaitTime: 30 >> >> # >> # Assuming that your outgoing call logic is kept in the # context called >> [outgoing] >> >> # Context: outgoing >> # Extension: 14109850123 >> # Priority: 1 >> >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> >> >> _______________________________________________ >> -- 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 >> > > > _______________________________________________ > -- 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 >
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