Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in advance!! Thanks, Max Alex Voip Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090407/62689f1d/attachment.htm
On Tue, 7 Apr 2009, Max Alex wrote:> Hi All, > I have working asterisk version 1.4.24. > I have a blind transfer issue with grandstream bt200. > I have updated the latest firmware to the phone. > The phone is sending the *refer* to asterisk but asterisk is not able to > connect with the another call > that i have checked in sip debug. > I am using transfer button of the grandstream phone. > Can anybody provide help for this issue? > Thanks in advance!!How are you doing the entire transfer operation? For blind transfers, I do: Push Transfer (caller is now on hold, you get a new dial-tone) dial extension and push SEND At this point, called phone rings and caller is immediately taken off hold and transfered to the new ringing phone... you can hang up at that point. Don't use the 'flash' key. I have many BT200's and GXP280's out there - this seems to work for them without any issues. Asterisk 1.2 though. Gordon
Max Alex wrote:> Hi All, > I have working asterisk version 1.4.24. > I have a blind transfer issue with grandstream bt200.Does it work with other phones? That means is it a Grandstream isue or a general issue?> I have updated the latest firmware to the phone. > The phone is sending the *refer* to asterisk but asterisk is not able to > connect with the another callWhy? some log messages would help us helping you.> that i have checked in sip debug. > I am using transfer button of the grandstream phone. > Can anybody provide help for this issue?Please ask again on the user mailing lists and provide some log messages> Thanks in advance!! > > Thanks, > Max Alex > Voip Developer > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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