I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that updates had been made to address this very problem, so I went to CVS HEAD and updated features.conf: [featuremap] blindxfer => *1 ; Blind transfer disconnect => *0 ; Disconnect atxfer => *2 ; Attended transfer Now, when a call comes in, I can press *1 and I hear "Transfer", at which point I enter an extension and the call goes there. However if _I_ initiate the call, *1 does nothing - I cannot transfer the call. Same story for attended transfer (*2). It doesn't make any difference whether I place the call on a SIP or ZAP channel. Is this a bug? If not, what's the secret to transferring outgoing calls that I initiate? BTW, I have calltransfer=yes on ALL my ZAP channels and I'm using t in my Dial commands (I noticed that using T doesn't help ? the called party can't transfer the call either). Thanks, Hugh
Mojo with Horan & Company, LLC
2005-Sep-27 12:18 UTC
[Asterisk-Users] blindxfer & atxfer not working?
double-check your usage of the t and T parameters to the Dial command, detailed here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Mojo hugolivude wrote:> I'm wondering whether there's a problem with the blindxfer and atxfer commands. > > I was using Asterisk STABLE and pressing the # key to transfer calls > worked fine, except of course when you called up FedEx and they asked > "Enter the number of packages, followed by the Pound key". > > I found on the wiki > (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) > that updates had been made to address this very problem, so I went to > CVS HEAD and updated features.conf: > > [featuremap] > blindxfer => *1 ; Blind transfer > disconnect => *0 ; Disconnect > atxfer => *2 ; Attended transfer > > Now, when a call comes in, I can press *1 and I hear "Transfer", at > which point I enter an extension and the call goes there. However if > _I_ initiate the call, *1 does nothing - I cannot transfer the call. > Same story for attended transfer (*2). > > It doesn't make any difference whether I place the call on a SIP or ZAP channel. > > Is this a bug? If not, what's the secret to transferring outgoing > calls that I initiate? > > BTW, I have calltransfer=yes on ALL my ZAP channels and I'm using t in > my Dial commands (I noticed that using T doesn't help ? the called > party can't transfer the call either). > > Thanks, > Hugh > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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-- Mojo <mojo@horanappraisals.com> Office Manger, Horan & Company, LLC (907) 747-6666 x112