Asterisk weirdness

["Mark Harrison (Groups)" <mph@xxxxxxxxxxxxxxx>]

I know that there are a lot of Asterisk users on the list, and figure
that internal phone systems with VOIP are on-topic these days.

My problem seems to be that calls don't terminate immediately, but
instead seem to time out.

I have the following hardware:

- an Asterisk server
- a Sipura SPA-2000 (connects two normal handsets to SIP)
- a Sipura SPA-841 (ethernet phone)

I have the following fragment from extensions.conf. The [SPA-2000-1]
refers to line 1 on the SPA-2000 (cunning, eh?). The SIP/marksphone
refers to the SPA-841, which sits in my study.

[SPA-2000-1]
exten =>21,1,Answer
exten =>21,2,Dial(SIP/marksphone,20,tr)
exten =>21,3,VoiceMail,u1234
exten =>21,103,VoiceMail,u1234


If I pick up the handset on line one of the SPA-2000-1, and dial
"21#",
[Note 1] then, sure enough, the phone in the study rings. Furthermore,
if someone picks it up, I can talk to them...

... once ...

If we both hang up, and then I redial, I get put straight through to the
voicemail, as if the SPA-841 weren't available.

If I then leave it 5 minutes (or restart *), then it works again.

I've not tested too scientifically, but the timeout period is "well in
excess of 30 seconds, but less than 5 minutes".

What am I missing from my config?

Regards,

Mark

[Note 1] - yup, I know, I need to sort out the dialling prefixes on the
SPA-2000 so I don't need to hit the "#" - that's on a "to
do" list once
I get the rest working. In the meantime, being able to NOT trigger
dial-through automatically is a useful debugging tool for
extensions.conf, IYSWIM.