Re: [asterisk-users] Call Transfer not working
On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
Re: [asterisk-users] Call Transfer not working
Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
Re: [asterisk-users] Call Transfer not working
Thanks everyone. I was using the Tt flag but in the wrong place in the dial application. Cheers On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima takehiro.dream...@gmail.com wrote: Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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
Re: [asterisk-users] Call Transfer not working
Hi. Maybe you forgotten specify to allow the transferring a call. Try with tT options in Dial() in extensions.conf. // I don't know what's difference t and T. -- Takehiro Matsushima takehiro.dream...@gmail.com 2012/4/7 Rizwan Hisham rizwanhas...@gmail.com: Hi All, I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf setting rfc2833 and inband. I have also enabled blind and attended transfer features in features.conf but still call transfers dont work. I have setup transfer feature in past but i dont think i am missing anything this time. I just dont have any clue why its not working. I have tried using ATAs and softphones but cant make it to work. Can anyone help? Am I missing anything? features show output: === Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # #1 Attended Transfer *2 One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor == -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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
Re: [asterisk-users] call transfer not working
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general settings. On 7/4/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --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 -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com ___ --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
[asterisk-users] call transfer not working
Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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
