[Asterisk-Users] Incoming SIP calls

[Brian Mulligan]

Hello All
I am trying to answer incoming SIP calls, first, by dialing an
extension, thence into voicemail, which works; and secondly by going
straight into voice mail which does not. The extension.conf that works
is like this;

[incomingSIP]
exten=>_.,1,Dial,Zap/2|1
exten=>_.,2,Voicemail,u5152
exten=>_.,3,Hangup

the extension.conf which does not is like this;

[incomingSIP]
exten=>_.,1,Answer
exten=>_.,2,Voicemail,u5152
exten=>_.,3,Hangup

For the non-working config I cam see the commands being run on the
console but the SIP session times out without receiving any audio. I
have traced both sessions with ethereal and the protocol handshake is
identical however * appears to be ignoring the ACK response for the
second config and repeatedly sends 200/OK and then times out.
Isuppose I am missing something obvious here but am going 'glassy eyed' 
trying to spot it.
Any help appreciated.
Brian