Hi all! I'm experimenting a strange problem in my Asterisk PBX: I've got an Asterisk pbx in the office: I dial an external number; the dialled number answers me correctly, but as soon as I press the '*' button (i.e. to navigate through the menus or to enter a password) my Asterisk box put me on hold.
(CLI transcription follows: -- Executing ChanIsAvail("SIP/222-23da", "Zap/g1&Zap/g2") in new stack -- Executing Cut("SIP/222-23da", "theChannel=AVAILCHAN||1") in new stack -- Executing NoOp("SIP/222-23da", "Zap/1") in new stack -- Executing Dial("SIP/222-23da", "Zap/1/34844503450||tTH") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/34844503450 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/222-23da [Here I press the '*' button] -- Started music on hold, class 'default', on Zap/1-1 -- Unable to find extension '' in context 'internal' -- Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (internal, 034844503450, 4) exited non-zero on 'SIP/222-23da') I think that Asterisk understands my postselection '*' DTMF tone like a command, not simply a tone to forward to the remote destination. How can I solve the problem? Tnx in advance Giovanni _______________________________________________ --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