asterisk users - Apr 2004 - SIP device rings once on busy before giving busy tone with dialplan

Hi!

I am having difficultly in having users of various SIP devices obtain the
correct behaviour when they call a busy number ie. only hearing the
Congestion/Busy tone.  I assume this might be because the SIP device
itself generates the 'ring' tone?

With my current setup in the dialplan extract (below) the user of the SIP
device hears one 'ring' and then the busy tone if a number is busy.

I have tried using 'Congestion' (instead of Answer+Hangup) but then
their
SIP phone rings indefinitely (or until the 45 secs timeout) and they never
know the number they called was busy and they wait needlessly for 45 secs.

I'm hoping in earnest that someone might be able to post back a quick
change to my dialplan to let me know if I can improve on this behaviour by
changing a few lines in the dialplan or a defining a macro or something or
just if this what happens to everyone with Asterisk perhaps you could let
me know so I can stop searching for a fix ?

I wrote this dialplan last October when I was really new to Asterisk and
just accepted the behaviour until now when I'm wondering if it can be
refined - I've tried and failed and read all that I could.

My dialplan is for the outgoing SIP call is:

exten => _00.,1,AbsoluteTimeout(3600)
exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten => _00.,3,Answer
exten => _00.,4,Hangup
exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten => _00.,104,Answer
exten => _00.,105,Hangup

(if call can go through on TRUNK1 send it out, if TRUNK1 is out of
capacity and therefore busy then try trunk 2 before giving up) if that is
busy (therefore it is likely the number really is busy then grab the
caller and hang them up (and they then hear 'busy').  

(using Congestion instead of Hangup gives the same behaviour here too but
the person calling out still gets the one 'ring' before the busy tone -
also removing the ,r makes no difference either)

Also, if TRUNK2 were busy would it be possible to go to TRUNK3 by
defining context 204 or not ?

Hoping someone has the patience to laugh at what I did and suggest how I
could fix it if it's fixable ?? maybe ? - please ?

Thanks for listening.  Have a great rest of weekend.

Chris

> My dialplan is for the outgoing SIP call is:
>
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
>
I can't help with presenting busy to the SIP devices, but if you have the
above on any sort of PSTN gateway you are going to annoy the PSTN users - as
if the number selected is busy or otherwise unavailable you will still
'Answer' the PSTN call, causing the person calling to pay whatever call
establishment charges/minimum charges appropriate to their tariff.

Linus

On Sat, 2004-04-17 at 15:58, Chris Orme wrote:
> My dialplan is for the outgoing SIP call is:
> 
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
> 
> (if call can go through on TRUNK1 send it out, if TRUNK1 is out of
> capacity and therefore busy then try trunk 2 before giving up) if that is
> busy (therefore it is likely the number really is busy then grab the
> caller and hang them up (and they then hear 'busy').  

Um - I'm probably missing the point entirely - but why are your trunks
not in a group and why are you not then using the group to dial out on?

(not posted to Asterisk - just you)
-- 
  .  .     ___. .__      Posix Systems - Sth Africa
 /| /|       / /__       mje@posix.co.za  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20040418/d4ecb7e8/attachment.pgp