Hi All, We have a 1.6.2.6 Asterisk box connected to a 1.2 asterisk box, when people dial to the conference room in 1.2 (from 1.6), of course they are prompted for a room number and flush it by dialing # sign, the problem when they hit #, this happened: -- Started music on hold, class 'default', on SIP/USPBX2-000007d5 -- <SIP/8425-000007d4> Playing 'pbx-transfer.gsm' (language 'en') and it gets disconnected. Anyone has a clue? Thank you! -- Khalid Touati -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120627/a0d158f3/attachment.htm>
That's fine guys I figured it out: under features.conf: [featuremap] ;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call! blindxfer => * I changed it to * and got rid of the pb -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120627/a29bc26e/attachment.htm>
On 6/27/2012 3:44 PM, khalid touati wrote:> #, this happened: > -- Started music on hold, class 'default', on SIP/USPBX2-000007d5 > -- <SIP/8425-000007d4> Playing 'pbx-transfer.gsm' (language 'en') > and it gets disconnected. Anyone has a clue?do you have # assigned in /etc/asterisk/features.conf ? perhaps to put the caller on hold ? -- Jeremy Kister http://jeremy.kister.net./