asterisk users - Mar 2003 - iconnecthere 480 error: is there a workaround?

I am going to have to find a fix for this problem or I'm going to have 
to quit using iconnect.

About one call in 10 or so, iconnect's gateway gives me an error 
(console output appended below).

So upon receiving the error, which as a 4XX error means, "Fatal," 
asterisk gives up and drops the call.  But not iconnect!!  The phone at 
the other end starts ringing, and rings several times before the call is 
dropped.

So the person at the other end, unless it's my friends who are now 
inured to this, wonder WTF is going on.

I sent a mail to iconnect asking if they don't agree that it's broken, 
but in the near-term I need to find a fix.

Thx.

B.

*****
Console output begins here, numbers elided to protect the innocent :-)

     -- Called 66661XXXNNNMMMM at iconnect
     -- Got SIP response 480 "Temporarily not available" back from 
213.137.73.140
   == No one is available to answer at this time
WARNING[311310]: File pbx.c, Line 1179 (ast_pbx_run): Channel 
'SIP/ata1-2da9' sent into invalid extension '66661XXXNNNMMMM' in
context
'iconn', but no invalid handler

I get these errors  (480 "Temporarily...") when I try to use my 
iconnect account quickly after hanging up on a previous session. 
They have some sort of contention locking system which allows only 
one call at a time on an account, and if you do not give it adequate 
time to "settle", you'll hit that error type.  I have found that 
waiting 15 seconds or so before making another call will ensure 
completion.

Personally, I think they should let multiple sessions through on the 
same account and have a "hard limit" set on consumablel minutes in a 
monthly billing period. In other words, if their concern is about 
fraud, then fine - make it such that the account holder must 
"recharge" their account past a certain limit.  Don't limit my 
burning of minutes due to poorly contrived fraud protection schemes; 
heck, you'd think they'd want customer to burn up minutes as quickly 
as possible.

JT


>I am going to have to find a fix for this problem or I'm going to 
>have to quit using iconnect.
>
>About one call in 10 or so, iconnect's gateway gives me an error 
>(console output appended below).
>
>So upon receiving the error, which as a 4XX error means, "Fatal," 
>asterisk gives up and drops the call.  But not iconnect!!  The phone 
>at the other end starts ringing, and rings several times before the 
>call is dropped.
>
>So the person at the other end, unless it's my friends who are now 
>inured to this, wonder WTF is going on.
>
>I sent a mail to iconnect asking if they don't agree that it's 
>broken, but in the near-term I need to find a fix.
>
>Thx.
>
>B.
>
>*****
>Console output begins here, numbers elided to protect the innocent :-)
>
>     -- Called 66661XXXNNNMMMM at iconnect
>     -- Got SIP response 480 "Temporarily not available" back from
>213.137.73.140
>   == No one is available to answer at this time
>WARNING[311310]: File pbx.c, Line 1179 (ast_pbx_run): Channel 
>'SIP/ata1-2da9' sent into invalid extension
'66661XXXNNNMMMM' in
>context 'iconn', but no invalid handler

I've found the same.

If I make an outgoing call (snom 200 handset), I get about 5 seconds
of audio and then it drops out (very occasionally it does work).

Incoming calls appear to work, though.

  -- Executing Goto("SIP/515-Office-143b",
"iconnecthere-ulaw|91800XXXXXXX|1") in new stack
  -- Goto (iconnecthere-ulaw,91800XXXXXXX,1)
  -- Executing StripMSD("SIP/515-Office-143b", "1") in new
stack
  -- Executing Dial("SIP/515-Office-143b",
"SIP/1800XXXXXXX@iconnecthere") in new stack
  -- Called 1800XXXXXXX@iconnecthere
  -- SIP/iconnecthere-960b answered SIP/515-Office-143b
  -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b
  -- Got SIP response 480 "Temporarily not available" back from
213.137.73.178
 == Spawn extension (iconnecthere-ulaw, 1800XXXXXXX, 2) exited non-zero on
'SIP/515-Office-143b'

SIP config is:

[general]
port=5060
bindaddr=0.0.0.0
context=sip-remote
disallow=all
allow=ulaw
allow=alaw
tos=lowdelay
tos=184
register => 1XXXXXXXXXX:XXXX@natrelay.deltathree.com

[iconnecthere]
type=friend
username=XXXXXXXX
password=XXXX
host=sipauth.deltathree.com
context=iconnecthere-ulaw
callerid="PADL Software Pty Ltd" <(XXX) XXX XXXX>
;txgain = 5.0;
;rxgain = 5.0;
inbanddtmf=1

-- Luke

P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As
I understand it, buying a LineJACK won't suffice if the card's DSP is
not actually used.
--
Luke Howard | PADL Software Pty Ltd | www.padl.com

>Is there a Record-Route header in the response that comes back from
>iconnect?
In the 480, not that I can see. 

In the 408, I'm not sure as I didn't have SIP debugging enabled (and I
don't
have anyone internationally to ring right now :-)).

-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.padl.com