We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the "Transfer" button on the phone (these phones are Grandstream BudgeTone 100s) or using the "#" key (the T/t flag must be set in the Dial command in asterisk for this way to work). Both methods seem similar; you enter the number and it transfers. The problems arise when the phone that it is transfered to is Busy or there is no answer: Asterisk just hangs up. Instead of this behaviour, we would like it to return the call to the person that transfered it. Alternativley, it could just do a 3 way call or something until the original person hangs up? I can't believe there is no way to achieve this. I have looked all over the internet but I can't find anything about this. From an Asterisk context point of view, the transfered call looks like a new call, and as far as i can see there is no way to differentiate between a new call and a transferred one. I know asterisk can tell the difference because the phone sends a "REFER" datagram to initate the transfer. Any help would be really appreciated Thanks, Alex Forrow Seek-it -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
I have the same problem and suppose that by some works on a new application similar to parkandannounce app. it should be done. - shabanip ----- Original Message ----- From: "Alex Forrow" <alex@seek-it.co.uk> To: <asterisk-users@lists.digium.com> Sent: Friday, September 17, 2004 5:43 PM Subject: [Asterisk-Users] Transferring Calls> We have set up an IP telephoney system hosted by Asterisk and its working > pretty well. We primarily use SIP and hardware IP phones. We have the > ability to transfer calls to another SIP phone using either the "Transfer" > button on the phone (these phones are Grandstream BudgeTone 100s) or using > the "#" key (the T/t flag must be set in the Dial command in asterisk for > this way to work). > > Both methods seem similar; you enter the number and it transfers. The > problems arise when the phone that it is transfered to is Busy or there is > no answer: Asterisk just hangs up. Instead of this behaviour, we would > like it to return the call to the person that transfered it. > > Alternativley, it could just do a 3 way call or something until the > original person hangs up? > > I can't believe there is no way to achieve this. I have looked all over > the internet but I can't find anything about this. > > From an Asterisk context point of view, the transfered call looks like a > new call, and as far as i can see there is no way to differentiate between > a new call and a transferred one. I know asterisk can tell the difference > because the phone sends a "REFER" datagram to initate the transfer. > > Any help would be really appreciated > > Thanks, > > > Alex Forrow > > Seek-it > > -- > Using Opera's revolutionary e-mail client: http://www.opera.com/m2/ > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Alex Forrow [mailto:alex@seek-it.co.uk] > Sent: Friday, September 17, 2004 9:13 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Transferring Calls > > > Both methods seem similar; you enter the number and it > transfers. The > problems arise when the phone that it is transfered to is > Busy or there is > no answer: Asterisk just hangs up. Instead of this behaviour, > we would > like it to return the call to the person that transfered it. >I am not sure that you can do exactly that, but using a stdexten macro you can give the caller the option to wait until the extension is available or go to voicemail.> Alternativley, it could just do a 3 way call or something until the > original person hangs up? >This is called an attended transfer and is very frequently a feature of the actual sip phone you are using. We are using this very Effectively with Cisco 7960's.> I can't believe there is no way to achieve this. I have > looked all over > the internet but I can't find anything about this. >Here are some pages that helped us work around this issue: * http://www.voip-info.org/wiki-Asterisk+tips+campon - Basically sets up an IVR menu that allows the caller to hit 1 to leave a message or hold on the line to get answered. * http://www.junghanns.net/asterisk/page6.html - Once set up it enables Follow me type of use. Then in your normal context execute the macro instead of Dial. Integrating these two features together has allowed us to accomplish the same goal. This even had an unexpected side effect for us over Our previous system which preformed like you wanted: it kept our Receptionists from dealing with the same call nearly doubling their effectiveness. I hope this helps, Robert Jackson