Sorry for bumping my own thread but just hoping that someone out there can help, I don't want to raise a bug on * if this is a strange issue with my dial plan
Just to clarify this is attended transfer using asterisk and not a phone feature (not joining two held calls etc) Could someone with 1.2.x give the following call flow a try? - Connect a call between two phones on * - Called party initiates attended transfer, e.g *1 depending on how your features.conf is setup. - Dial internal extension - Ringing from transfer extension - Hangup original called party *I / most people would expect the caller to be connected to the ringing extension but instead on my * they get disconnected. In other words in this scenario users expect the attended transfer to switch to the same call flow of blind transfer. I would look into the code but am a Java / PHP dev :( Thank you for your help Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
