[Asterisk-Users] 'I'nvalid extension handling problems, even with workaround

[telmo at n1.com]

Hello folks,

I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.

First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
    http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i' extension can't be used like I (and
apparently the wiki page author) thought.

I then did the "separate context with _." trick the above wiki page suggests; at
first it seemed to work: picking up an extension and dialing any invalid
extension would play the message (albeit it would play twice, can't understand
why) and then hang up.

Later I found the above configuration was interfering with my sip dial-out thru
voiptalk.org: any call I place thru voiptalk (for example, dialing '8902' for the
welcome message) is followed by the "invalid extension" message when the remote
end hangs up.

I'm running Asterisk 1.0.2, which I compiled from the source myself.
Below are my extensions.conf and sip.conf, simplified to the point where there
isn't anything not related to the above problem:

;;;extensions.conf
[internal]              ;;; context used by our internal SIP-phon
include => voiptalk.org         ;include context below
exten => 11,1,Dial(SIP/gsbt100,20,tr);calling <11>: dial our office phone
include => invalid_calls        ;all ext numbers not handled above are invalid
[voiptalk.org]
;forwards any calls starting with an "8" thru voiptalk.org
exten => _8.,1,Answer
exten => _8.,3,SetCIDNum(55555555)
exten => _8.,4,SetCIDName(My Name And Surname)
exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
exten => _8.,6,HangUp
[invalid_calls]         ;;; default context for invalid calls
exten => _.,1,Wait(1)
exten => _.,2,Answer
exten => _.,3,Playback(invalid)
exten => _.,4,Hangup
;;;end of extensions.conf

;;;sip.conf
[general]
register => 55555555:7777777 at voiptalk.org/1000
[voiptalk.org]
type=friend
secret=7777777
username=55555555
host=voiptalk.org
fromdomain=voiptalk.org
insecure=very
dtmfmode=info
[gsbt100]                                                      
type=friend                                                    
host=dynamic
defaultip=192.168.1.200
canreinvite=no
username=gsbt100
secret=xpto1234
dtmfmode=rfc2833
mailbox=1234
context=internal
callerid="Office" <11>
;;;eof sip.conf

Below is the console output of 'asterisk -cp -vvvvvvvvvvvvvv', when I pick up the
phone on the gsbt100 and dial '8902': 

Setting NAT on RTP to 0
Stopping retransmission on '20fc1bc647b262ea at 192.168.1.200' of Response 15995: Found
Setting NAT on RTP to 0
Check for res for gsbt100
Call from user 'gsbt100' is 1 out of 0
build_route: Contact hop: <sip:gsbt100 at 192.168.1.200;user=phone>
    -- Executing Answer("SIP/gsbt100-b25b", "") in new stack
    -- Executing Answer("SIP/gsbt100-b25b", "") in new stack
    -- Executing SetCIDNum("SIP/gsbt100-b25b", "55555555") in new stack
    -- Executing SetCIDName("SIP/gsbt100-b25b", "My Name And Surname") in new stack
    -- Executing Dial("SIP/gsbt100-b25b", "SIP/902 at voiptalk.org|999|g") in new stack
SIMPLE DIAL (NO URL)
Setting NAT on RTP to 0
Outgoing Call for 902
902 is not a local user
    -- Called 902 at voiptalk.org
Stopping retransmission on '20fc1bc647b262ea at 192.168.1.200' of Response 15996: Found
(Provisional) Stopping retransmission (but retaining packet) on
'645df6fa2d6126026dd9624d5c3f35e1 at voiptalk.org' Request 102: Found
Acked pending invite 102
Stopping retransmission on '645df6fa2d6126026dd9624d5c3f35e1 at voiptalk.org' of
Request 102: Found
(Provisional) Stopping retransmission (but retaining packet) on
'645df6fa2d6126026dd9624d5c3f35e1 at voiptalk.org' Request 103: Found
Acked pending invite 103
Stopping retransmission on '645df6fa2d6126026dd9624d5c3f35e1 at voiptalk.org' of
Request 103: Found
build_route: Record-Route hop: <sip:902 at 82.145.32.73;ftag=as2be826f1;lr=on>
build_route: Contact hop: <sip:902 at 217.14.132.184>
    -- SIP/voiptalk.org-8789 answered SIP/gsbt100-b25b
    -- Attempting native bridge of SIP/gsbt100-b25b and SIP/voiptalk.org-8789
Oooh, format changed to 8
Ooh, format changed from UNKN to ULAW
Ooh, format changed from UNKN to ALAW
Didn't get a frame from channel: SIP/voiptalk.org-8789
Bridge stops bridging channels SIP/gsbt100-b25b and SIP/voiptalk.org-8789
update_user_counter(902) - decrement outUse counter
902 is not a local user
Exiting with DIALSTATUS=ANSWER.
    -- Executing Hangup("SIP/gsbt100-b25b", "") in new stack
  == Spawn extension (internal, 8902, 6) exited non-zero on 'SIP/gsbt100-b25b'
    -- Executing Wait("SIP/gsbt100-b25b", "1") in new stack
    -- Executing Answer("SIP/gsbt100-b25b", "") in new stack
    -- Executing Playback("SIP/gsbt100-b25b", "invalid") in new stack
Difference is 50232, ms is 6299
    -- Playing 'invalid' (language 'en')
Request to schedule in the past?!?!
Request to schedule in the past?!?!
    -- Executing Hangup("SIP/gsbt100-b25b", "") in new stack
  == Spawn extension (internal, h, 4) exited non-zero on 'SIP/gsbt100-b25b'
update_user_counter(gsbt100) - decrement inUse counter
Stopping retransmission on '20fc1bc647b262ea at 192.168.1.200' of Request 102: Found

What happens on the phone is that I hear voiptalk.org's greeting and after they
hang up, I hear my own "invalid extensions" message.

I've searched the wiki, the list archives and even bugs.digium.com for an answer,
but could not find any. Thanks in advance for your help.

Best Regards,
   Telmo.

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