2011/10/5 Nasir Iqbal <[email protected]> > You can do this by an AMI based transfer (Redirect) to Local channel, and > then in local channel's dialplan you need to add your desired custom sip > header followed by a dial command. > > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.com/ > > Thanks for the tip : I'll give it a try ASAP (and report here)! Thanks again !
> > > On Wed, Oct 5, 2011 at 11:36 AM, Olivier <[email protected]> wrote: > >> >> 2011/10/4 Olivier <[email protected]> >> >>> Hi, >>> >>> Has anyone heard (or read) about an existing or emerging standard >>> targeting the following feature : >>> 1. a SIP handset receives an incoming call >>> 2. this handset starts ringing >>> 3. then it receives an update asking to autoanswer the ringing call. >>> >>> This feature would help to build software panels complementing or >>> replicating hard phones GUI. >>> >>> (I know you can work around such feature using conference rooms or >>> dealing with hard phones API (really ?) but in order to keep Queue log >>> accurate, this feature would be useful). >>> >>> Cheers >>> >> >> Hi, >> >> In my quest to allow a software panel to ask an hardphone to answer an >> incoming call without touching the hardphone itself, I'm wondering if a >> Reinvite application could exist. >> >> I'm thinking about the following use case : >> >> Alice is calling Bob >> Bob's phone starts to ring >> Bob's GUI app also shows the incoming call asking him if he prefers to >> reject, transfer or answer the call >> With the GUI app, Bob replies he whishes to reply >> Asterisk reinvites Bob's phone with autoanswer option >> Bob's phone answers Alice phone >> >> Thoughts ? >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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