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telmo@n1.com

2004-Dec-20 02:20 UTC

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

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@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@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@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@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@voiptalk.org
Stopping retransmission on '20fc1bc647b262ea@192.168.1.200' of Response
15996: Found
(Provisional) Stopping retransmission (but retaining packet) on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' Request 102: Found
Acked pending invite 102
Stopping retransmission on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' of
Request 102: Found
(Provisional) Stopping retransmission (but retaining packet) on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' Request 103: Found
Acked pending invite 103
Stopping retransmission on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' of
Request 103: Found
build_route: Record-Route hop:
<sip:902@82.145.32.73;ftag=as2be826f1;lr=on>
build_route: Contact hop: <sip:902@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@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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telmo@n1.com

2004-Dec-22 02:40 UTC

head link

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

Hello folks,

Humrmrm... 2 days, no answers... :-/

Either I made a stupid question (I don't think so: I have *really* tried to
solve
that on my own before asking the list) or this one's just something nobody
has
ever tried but me (I also find that unlikely: even the telco here plays a
message
when I dial a wrong number; also there's the wiki page I mentioned, which
indicates that someone in the past has had the same issue).

It may also be that this one worked perfectly with previous versions of
Asterisk,
and isn't working with the version I'm trying (this could be more
probable: I've
seen people reporting many broken things in 1.0.2 lately).

Could anyone please clarify me? What are folks using for "wrong
extension" handling?

Thanks in advance for any answers.

Best Regards,
Telmo.


On Mon Dec 20  1:20 , telmo@n1.com> sent:
>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 : 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@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" 
>;;;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@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: 
>    -- 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@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@voiptalk.org
>Stopping retransmission on '20fc1bc647b262ea@192.168.1.200' of
Response 15996: Found
>(Provisional) Stopping retransmission (but retaining packet) on
>'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' Request 102: Found
>Acked pending invite 102
>Stopping retransmission on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' of
>Request 102: Found
>(Provisional) Stopping retransmission (but retaining packet) on
>'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' Request 103: Found
>Acked pending invite 103
>Stopping retransmission on
'645df6fa2d6126026dd9624d5c3f35e1@voiptalk.org' of
>Request 103: Found
>build_route: Record-Route hop: 
>build_route: Contact hop: 
>    -- 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@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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telmo@n1.com

2004-Dec-22 20:27 UTC

head link

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

Hello Rich,

First of all, thank you very much for your help and patience.

I've just arrived home from work (yes, I'm one of the midnight oil
burners :-))
and implemented and tested your suggestions; unfortunatelly it didn't work,
the
same behaviour remains.

More details follow below, in-line:

On Wed Dec 22  4:14 , Rich Adamson <radamson@routers.com>
sent:>Inline...
>
>> Humrmrm... 2 days, no answers... :-/
>
>Well, let me see if I can take a stab at this one.
You sure did. Your help was most appreciated, I learned a lot from thinking
about
your suggestions, even if they did not work as planned (see below) they made a
lot of sense.
>After working with * for about a year now, I'd suggest the toughest
>part of the learning curve is truly understanding how to take advantage
>of the various 'context' statements to accomplish an objective. Part
of
>reason for the steep learning curve appears to relate to the lack of
>any reasonable form of tracing/debugging what the system is actually
>using for a context at each step. (What I mean is that its not intutive
>for the beginner.)
I agree. I'm working with 15 (!) "-v" on the command-line here and
yet I can only
perceive Asterisk has left or entered a context indirectly, by the commands that
are executing.
>It would really be nice if there was a 'debug context' type command
>that would simply display each extensions.conf line as it is executed.
That can't be too hard to implement, when I'm a little more familiarized
with
Asterisk I will certainly try to code that in.
>> Either I made a stupid question (I don't think so: I have *really*
tried to solve
>> that on my own before asking the list) or this one's just something
nobody has
>> ever tried but me (I also find that unlikely: even the telco here plays
a message
>> when I dial a wrong number; also there's the wiki page I mentioned,
which
>> indicates that someone in the past has had the same issue).
>> 
>> >I'm having trouble configuring Asterisk to play an
"invalid extension" message to
>> >anyone dialing an undefined extension.
>> 
>> >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.
>> 
>> >;;;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 : dial our office
phone
>> >include => invalid_calls        ;all ext numbers not handled
above are invalid
>
>The 'separate context' approach _does_ work, but you've just
confused
>that approach by dropping the Dial statement in the middle. Change this
>to something like:
> [internal]
> include => valid-extensions
> include => voiptalk
> include => any-other-context-that-you-need
> include => invalid-calls
Makes a lot of sense, and also leads to much more intelligible structure. I
implemented it almost literally as you suggest (see below for my extensions.conf
file after the modifications).
>The above _sequence_ of include statements is maintained for each call.
>In other words, if a call does _not_ match entries in
'valid-extensions',
>then it proceeds to the next include. However, if a match is found
(including
>special cases such as 't', etc) then the call processing may _not_
step
>through the remaining includes.
OK. I understand what you are stating, it makes a lot of sense. Unfortunatelly
it
seems Asterisk disagrees with us... :-) please see below.
>I'm not sure at all about using context names with a period in it, so
>just to ensure that isn't causing an issue, stick to context names with
>alphanumeric characters. (At least eliminate that uncertainty.)
OK. Let's play it _really_ safe: I've removed "." from context
names, and
replaced "_" (underline) for "-" (dashes), so right now
I'm only using characters
from the set [a-z0-9-] in context names.
>> >[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
>
>In the above [invalid_calls] context, change the order to:
> exten => _.,1,Answer
> exten => _.,2,Wait(1)
> exten => _.,3,Playback(invalid)
> exten => _.,4,Hangup
Also makes a lot of sense. Done it that way.
>It is rather important from a learning perspective that you create
>the [internal] context for use by "internal phones only".
Don't try
>to use that same context for incoming voiptalk calls, etc. For incoming
>external calls, create a separate context something like:
>[incoming-voiptalk]
> include => valid-extensions
> include => invalid-calls
Ah! Thanks for the tip. Also makes a lot of sense.
>After you've made the above changes, "stop" and restart
asterisk to
>ensure you know exactly what asterisk believes the dialplan is supposed
>to be. Stay away from the various 'reload' commands until you
understand
>exactly what is happening.
Did that: "stop now" in the cli, and then restarted asterisk with
"asterisk -cp
-vvvvvvvvvvvvvv" after modifying the files.
>Then place test calls and watch the CLI paying close attention to which
>contexts are actually in use (or which context a call stops progressing
>in). If necessary, increase the debug level to see the needed call progress
>info.
Did it; unfortunatelly the same behaviour continues: when I call (for example)
8902, I hear Voiptalk's welcome message and right after it I hear the
"invalid
extension" message...

The files ended up as below:

;;;extensions.conf
[internal]
include => valid-extensions     ;; internal sip phones
include => voiptalk             ;; dial out thru voiptalk.org
include => tests                ;; miscellaneous tests
include => invalid-calls        ;all ext numbers not handled above are
invalid
[valid-extensions]              ;;; context used by our internal SIP-phon
exten => 11,1,Dial(SIP/gsbt100,20,tr)   ;calling <11>: dial our office
phone
[voiptalk]
;forwards any calls starting with an "8" thru voiptalk.org
exten => _8.,1,Answer
exten => _8.,3,SetCIDNum(55555555)
exten => _8.,4,SetCIDName(My Full Name)
exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk,999,g)
exten => _8.,6,HangUp
[invalid-calls]         ;;; default context for invalid calls
exten => _.,1,Answer
exten => _.,2,Wait(1)
exten => _.,3,Playback(invalid)
exten => _.,4,Hangup
;;;eof extensions.conf


;;;sip.conf
[general]
register => 55555555:666666@voiptalk.org/1000
maxexpirey = 180
defaultexpirey = 160
[voiptalk]
type=friend
secret=666666
username=55555555
host=voiptalk.org
fromdomain=voiptalk.org
insecure=very  ;needed to allow incoming calls to bypass authentication
dtmfmode=info   ;Choices are inband, rfc2833, or info
[gsbt100]
type=friend
host=dynamic            ;must use dynamic even if fixed ip/hostname, or else
defaultip=192.168.1.200 ;asterisk won't allow it to register
canreinvite=no          ;avoid trouble with NAT: keeps Asterisk "in the
middle"
username=gsbt100        ;gsbt100 sends that when the respective fields are empty
secret=mysecret
dtmfmode=rfc2833        ;Choices are inband, rfc2833, or info
mailbox=1234            ;Mailbox for message waiting indicator
context=internal        ;Context for internal SIP phones
callerid="Office" <11>  ;format is "name"
<number>
;;;eof sip.conf

Here comes the output on the console when I dial '8902':

Setting NAT on RTP to 0
Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of Response
45199: F
ound
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@192.168.1.200;user=phone>
    -- Executing Answer("SIP/gsbt100-7dcf", "") in new stack
    -- Executing Wait("SIP/gsbt100-7dcf", "1") in new stack
Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of Response
45200: F
ound
    -- Executing SetCIDNum("SIP/gsbt100-7dcf", "55555555")
in new stack
    -- Executing SetCIDName("SIP/gsbt100-7dcf", "My Full
Name") in new stack
    -- Executing Dial("SIP/gsbt100-7dcf",
"SIP/902@voiptalk|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@voiptalk
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' Request 102: Found
Acked pending invite 102
Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
Request 102: Found
Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
Request 102: Not Found
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' Request 103: Found
Acked pending invite 103
Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
Request 103: Found
build_route: Record-Route hop:
<sip:902@82.145.32.73;ftag=as3688ce90;lr=on>
build_route: Contact hop: <sip:902@217.14.132.184>
    -- SIP/voiptalk-bbd4 answered SIP/gsbt100-7dcf
    -- Attempting native bridge of SIP/gsbt100-7dcf and SIP/voiptalk-bbd4
Oooh, format changed to 8
Ooh, format changed from UNKN to ULAW
Ooh, format changed from UNKN to ALAW
Registration from '<sip:gsht286@192.168.1.1;user=phone>' failed
for '192.168.1.201'
Auto destroying call 'd761b1512a710969@192.168.1.201'
Scheduled a timeout # 55
Stopping retransmission on
'327b23c6643c98696633487374b0dc51@192.168.1.1' of
Request 107: Found
Stopping retransmission on
'327b23c6643c98696633487374b0dc51@192.168.1.1' of
Request 108: Found
Registration successful
Cancelling timeout 55
Auto destroying call '8a7ae46c9e0d1370@192.168.1.201'
Didn't get a frame from channel: SIP/voiptalk-bbd4
Bridge stops bridging channels SIP/gsbt100-7dcf and SIP/voiptalk-bbd4
update_user_counter(902) - decrement outUse counter
902 is not a local user
Exiting with DIALSTATUS=ANSWER.
    -- Executing Hangup("SIP/gsbt100-7dcf", "") in new stack
  == Spawn extension (internal, 8902, 6) exited non-zero on
'SIP/gsbt100-7dcf'
    -- Executing Answer("SIP/gsbt100-7dcf", "") in new stack
    -- Executing Wait("SIP/gsbt100-7dcf", "1") in new stack
    -- Executing Playback("SIP/gsbt100-7dcf", "invalid") in
new stack
Difference is 50272, ms is 6304
    -- Playing 'invalid' (language 'en')
    -- Executing Hangup("SIP/gsbt100-7dcf", "") in new stack
  == Spawn extension (internal, h, 4) exited non-zero on
'SIP/gsbt100-7dcf'
update_user_counter(gsbt100) - decrement inUse counter
Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of Request
102: Found

As I said above, what I hear on the gsbt100 phone when I pick it up and dial
'8902' remains unchanged: first I hear VoipTalk's message, and
it's followed by
my "invalid" message. Examining the console output above seems to 
indicate that
Asterisk is simply "falling thru" the list of contexts included in the
"[internal]" context, contrarywise to what we were expecting...

As I said, I'm using Asterisk 1.0.2; Can this be a bug in that release? If
so,
can you indicate me a release where you think this works?

Thanks again for your help.

Best Regards,
   Telmo.


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telmo@n1.com

2004-Dec-24 09:53 UTC

head link

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

Hi Rich,

Thanks for the further help you are extending me. I will be trying your
suggestions during the holidays, and when I get back on-line will post my
progress here.

Thanks again.

Merry Christmas to all,
-- 
   Telmo.



On Thu Dec 23  5:06 , Rich Adamson  <radamson@routers.com> sent:
>> Hello Rich,
>> 
>> First of all, thank you very much for your help and patience.
>> 
>> I've just arrived home from work (yes, I'm one of the midnight
oil burners :-))
>> and implemented and tested your suggestions; unfortunatelly it
didn't work, the
>> same behaviour remains.
>> 
>> More details follow below, in-line:
>> 
>> On Wed Dec 22  4:14 , Rich Adamson radamson@routers.com> sent:
>> >Inline...
>> >
>> >> Humrmrm... 2 days, no answers... :-/
>> >
>> >Well, let me see if I can take a stab at this one.
>> 
>> You sure did. Your help was most appreciated, I learned a lot from
thinking about
>> your suggestions, even if they did not work as planned (see below) they
made a
>> lot of sense.
>> 
>> >After working with * for about a year now, I'd suggest the
toughest
>> >part of the learning curve is truly understanding how to take
advantage
>> >of the various 'context' statements to accomplish an
objective. Part of
>> >reason for the steep learning curve appears to relate to the lack
of
>> >any reasonable form of tracing/debugging what the system is
actually
>> >using for a context at each step. (What I mean is that its not
intutive
>> >for the beginner.)
>> 
>> I agree. I'm working with 15 (!) "-v" on the command-line
here and yet I can only
>> perceive Asterisk has left or entered a context indirectly, by the
commands that
>> are executing.
>> 
>> >It would really be nice if there was a 'debug context' type
command
>> >that would simply display each extensions.conf line as it is
executed.
>> 
>> That can't be too hard to implement, when I'm a little more
familiarized with
>> Asterisk I will certainly try to code that in.
>> 
>> >> Either I made a stupid question (I don't think so: I have
*really* tried to
solve>> >> that on my own before asking the list) or this one's just
something nobody has
>> >> ever tried but me (I also find that unlikely: even the telco
here plays a
message>> >> when I dial a wrong number; also there's the wiki page I
mentioned, which
>> >> indicates that someone in the past has had the same issue).
>> >> 
>> >> >I'm having trouble configuring Asterisk to play an
"invalid extension"
message to>> >> >anyone dialing an undefined extension.
>> >> 
>> >> >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.
>> >> 
>> >> >;;;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 : dial
our office phone
>> >> >include => invalid_calls        ;all ext numbers not
handled above are invalid
>> >
>> >The 'separate context' approach _does_ work, but you've
just confused
>> >that approach by dropping the Dial statement in the middle. Change
this
>> >to something like:
>> > [internal]
>> > include => valid-extensions
>> > include => voiptalk
>> > include => any-other-context-that-you-need
>> > include => invalid-calls
>> 
>> Makes a lot of sense, and also leads to much more intelligible
structure. I
>> implemented it almost literally as you suggest (see below for my
extensions.conf
>> file after the modifications).
>> 
>> >The above _sequence_ of include statements is maintained for each
call.
>> >In other words, if a call does _not_ match entries in
'valid-extensions',
>> >then it proceeds to the next include. However, if a match is found
(including
>> >special cases such as 't', etc) then the call processing
may _not_ step
>> >through the remaining includes.
>> 
>> OK. I understand what you are stating, it makes a lot of sense.
Unfortunatelly it
>> seems Asterisk disagrees with us... :-) please see below.
>> 
>> >I'm not sure at all about using context names with a period in
it, so
>> >just to ensure that isn't causing an issue, stick to context
names with
>> >alphanumeric characters. (At least eliminate that uncertainty.)
>> 
>> OK. Let's play it _really_ safe: I've removed "."
from context names, and
>> replaced "_" (underline) for "-" (dashes), so right
now I'm only using characters
>> from the set [a-z0-9-] in context names.
>> 
>> >> >[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
>> >
>> >In the above [invalid_calls] context, change the order to:
>> > exten => _.,1,Answer
>> > exten => _.,2,Wait(1)
>> > exten => _.,3,Playback(invalid)
>> > exten => _.,4,Hangup
>> 
>> Also makes a lot of sense. Done it that way.
>> 
>> >It is rather important from a learning perspective that you create
>> >the [internal] context for use by "internal phones only".
Don't try
>> >to use that same context for incoming voiptalk calls, etc. For
incoming
>> >external calls, create a separate context something like:
>> >[incoming-voiptalk]
>> > include => valid-extensions
>> > include => invalid-calls
>> 
>> Ah! Thanks for the tip. Also makes a lot of sense.
>> 
>> >After you've made the above changes, "stop" and
restart asterisk to
>> >ensure you know exactly what asterisk believes the dialplan is
supposed
>> >to be. Stay away from the various 'reload' commands until
you understand
>> >exactly what is happening.
>> 
>> Did that: "stop now" in the cli, and then restarted asterisk
with "asterisk -cp
>> -vvvvvvvvvvvvvv" after modifying the files.
>> 
>> >Then place test calls and watch the CLI paying close attention to
which
>> >contexts are actually in use (or which context a call stops
progressing
>> >in). If necessary, increase the debug level to see the needed call
progress
>> >info.
>> 
>> Did it; unfortunatelly the same behaviour continues: when I call (for
example)
>> 8902, I hear Voiptalk's welcome message and right after it I hear
the "invalid
>> extension" message...
>> 
>> The files ended up as below:
>> 
>> ;;;extensions.conf
>> [internal]
>> include => valid-extensions     ;; internal sip phones
>> include => voiptalk             ;; dial out thru voiptalk.org
>> include => tests                ;; miscellaneous tests
>> include => invalid-calls        ;all ext numbers not handled above
are invalid
>> [valid-extensions]              ;;; context used by our internal
SIP-phon
>> exten => 11,1,Dial(SIP/gsbt100,20,tr)   ;calling : dial our office
phone
>> [voiptalk]
>> ;forwards any calls starting with an "8" thru voiptalk.org
>> exten => _8.,1,Answer
>> exten => _8.,3,SetCIDNum(55555555)
>> exten => _8.,4,SetCIDName(My Full Name)
>> exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk,999,g)
>> exten => _8.,6,HangUp
>> [invalid-calls]         ;;; default context for invalid calls
>> exten => _.,1,Answer
>> exten => _.,2,Wait(1)
>> exten => _.,3,Playback(invalid)
>> exten => _.,4,Hangup
>> ;;;eof extensions.conf
>> 
>> 
>> ;;;sip.conf
>> [general]
>> register => 55555555:666666@voiptalk.org/1000
>> maxexpirey = 180
>> defaultexpirey = 160
>> [voiptalk]
>> type=friend
>> secret=666666
>> username=55555555
>> host=voiptalk.org
>> fromdomain=voiptalk.org
>> insecure=very  ;needed to allow incoming calls to bypass authentication
>> dtmfmode=info   ;Choices are inband, rfc2833, or info
>> [gsbt100]
>> type=friend
>> host=dynamic            ;must use dynamic even if fixed ip/hostname, or
else
>> defaultip=192.168.1.200 ;asterisk won't allow it to register
>> canreinvite=no          ;avoid trouble with NAT: keeps Asterisk
"in the middle"
>> username=gsbt100        ;gsbt100 sends that when the respective fields
are empty
>> secret=mysecret
>> dtmfmode=rfc2833        ;Choices are inband, rfc2833, or info
>> mailbox=1234            ;Mailbox for message waiting indicator
>> context=internal        ;Context for internal SIP phones
>> callerid="Office"   ;format is "name" 
>> ;;;eof sip.conf
>> 
>> Here comes the output on the console when I dial '8902':
>> 
>> Setting NAT on RTP to 0
>> Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of
Response 45199: F
>> ound
>> Setting NAT on RTP to 0
>> Check for res for gsbt100
>> Call from user 'gsbt100' is 1 out of 0
>> build_route: Contact hop: 
>>     -- Executing Answer("SIP/gsbt100-7dcf", "") in
new stack
>>     -- Executing Wait("SIP/gsbt100-7dcf", "1") in
new stack
>> Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of
Response 45200: F
>> ound
>>     -- Executing SetCIDNum("SIP/gsbt100-7dcf",
"55555555") in new stack
>>     -- Executing SetCIDName("SIP/gsbt100-7dcf", "My Full
Name") in new stack
>>     -- Executing Dial("SIP/gsbt100-7dcf",
"SIP/902@voiptalk|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@voiptalk
>> (Provisional) Stopping retransmission (but retaining packet) on
>> '12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' Request 102:
Found
>> Acked pending invite 102
>> Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
>> Request 102: Found
>> Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
>> Request 102: Not Found
>> (Provisional) Stopping retransmission (but retaining packet) on
>> '12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' Request 103:
Found
>> Acked pending invite 103
>> Stopping retransmission on
'12e2f2cc30702a0c3647cbeb4d2d2193@voiptalk.org' of
>> Request 103: Found
>> build_route: Record-Route hop: 
>> build_route: Contact hop: 
>>     -- SIP/voiptalk-bbd4 answered SIP/gsbt100-7dcf
>>     -- Attempting native bridge of SIP/gsbt100-7dcf and
SIP/voiptalk-bbd4
>> Oooh, format changed to 8
>> Ooh, format changed from UNKN to ULAW
>> Ooh, format changed from UNKN to ALAW
>> Registration from '' failed for '192.168.1.201'
>> Auto destroying call 'd761b1512a710969@192.168.1.201'
>> Scheduled a timeout # 55
>> Stopping retransmission on
'327b23c6643c98696633487374b0dc51@192.168.1.1' of
>> Request 107: Found
>> Stopping retransmission on
'327b23c6643c98696633487374b0dc51@192.168.1.1' of
>> Request 108: Found
>> Registration successful
>> Cancelling timeout 55
>> Auto destroying call '8a7ae46c9e0d1370@192.168.1.201'
>> Didn't get a frame from channel: SIP/voiptalk-bbd4
>> Bridge stops bridging channels SIP/gsbt100-7dcf and SIP/voiptalk-bbd4
>> update_user_counter(902) - decrement outUse counter
>> 902 is not a local user
>> Exiting with DIALSTATUS=ANSWER.
>>     -- Executing Hangup("SIP/gsbt100-7dcf", "") in
new stack
>>   == Spawn extension (internal, 8902, 6) exited non-zero on
'SIP/gsbt100-7dcf'
>>     -- Executing Answer("SIP/gsbt100-7dcf", "") in
new stack
>>     -- Executing Wait("SIP/gsbt100-7dcf", "1") in
new stack
>>     -- Executing Playback("SIP/gsbt100-7dcf",
"invalid") in new stack
>> Difference is 50272, ms is 6304
>>     -- Playing 'invalid' (language 'en')
>>     -- Executing Hangup("SIP/gsbt100-7dcf", "") in
new stack
>>   == Spawn extension (internal, h, 4) exited non-zero on
'SIP/gsbt100-7dcf'
>> update_user_counter(gsbt100) - decrement inUse counter
>> Stopping retransmission on '5e5172baa8c28adf@192.168.1.200' of
Request 102: Found
>> 
>> As I said above, what I hear on the gsbt100 phone when I pick it up and
dial
>> '8902' remains unchanged: first I hear VoipTalk's message,
and it's followed by
>> my "invalid" message. Examining the console output above
seems to  indicate that
>> Asterisk is simply "falling thru" the list of contexts
included in the
>> "[internal]" context, contrarywise to what we were
expecting...
>> 
>> As I said, I'm using Asterisk 1.0.2; Can this be a bug in that
release? If so,
>> can you indicate me a release where you think this works?
>
>Comments relative to the above:
>
>1. Without knowing exactly how your * box is configured, it would appear
>you are using some sort of firewall/nat box. The [voiptalk] context in
>sip.conf does not include anything relating to nat=yes, etc. Might look
>a little closer at the nat requirements.
>
>2. Not sure it will make any significant difference, but you might add
>canreinvite=no to the [voiptalk] context.
>
>3. I see addresses like 192.168.1.200, 1.201, and 1.1 in the above. What
>is 192.168.1.1?
>
>4. It would appear there are many more definitions in extensions.conf
>that you've chosen to not show us, and some of these might be important
>towards diagnosing the problem. Example, I see:
> == Spawn extension (internal, h, 4) exited non-zero on
'SIP/gsbt100-7dcf
>in the CLI stuff, but there config files you shown don't include an
"h"
>section in the internal context. Have you defined that? (If you don't
>want to post your entire sip.conf and extensions.conf, then at least email
>direct (minus secrets) to me.
>
>5. The syntax of:
> exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk,999,g)
>does not appear to be correct. Try something like:
> exten => _8.,5,Dial(Sip/voiptalk/${EXTEN:1}
>where 'voiptalk' is referring to the context [voiptalk] and not a
domain
>name. (Check the exact syntax as I typed the above from memory.)
>You said you dialed 8902 in the example above. But, the debug shows:
>  -- Executing Dial("SIP/gsbt100-7dcf",
"SIP/902@voiptalk|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@voiptalk
>Is extension 902 at voiptalk a valid extension? (Doesn't look like it to
me.)
>
>I'm more then a little confused with the above. It would almost appear 
>that you are expecting the [invalid-calls] context to be executed when
>an invalid extension "at" voiptalk is attempted. If so, that
isn't the
>way this stuff works. The [invalid-calls] context will only get executed
>when there is no matching dialplan entries on your * box (not theirs),
>and 8902 found a match in the form of _8. in the [voiptalk] context.
>
>6. Your voiptalk register command looks like:
> register => 55555555:666666@voiptalk.org/1000
>which tells the voiptalk.org folks to always send incoming calls to your
>* box to extension 1000. But, your extensions.conf doesn't include an
>entry for exten => 1000,1,...
>Is that because it was left out of the copy/paste, or does it not really
>exist?
>
>7. In the sip.conf definitions, definitions that use type=friend do not
>require/want any statements associated with an IP address. The registration
>process 'supplies' the appropriate IP address to asterisk. If the
>registration process is not working without those statements, then debug
>that before trying to debug the other stuff.
>
>Rich
>
>


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Nabeel Jafferali

2005-Jan-03 22:03 UTC

head link

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

> 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.
That's because your _X. extension has a 2nd line. Reduce it to
_X.,1,Playback(invalid) and you won't have that problem.

-- 
Nabeel Jafferali
tel: 416.491.9136 (toronto)
     646.225.7426 (new york)
fwd: 46990
email/msn : nabeel<at>jafferali.net

Nabeel Jafferali

2005-Jan-04 08:44 UTC

head link

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

asterisk-users-bounces@lists.digium.com wrote:> > > 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. 
> > 
> > That's because your _X. extension has a 2nd line. Reduce it to
> > _X.,1,Playback(invalid) and you won't have that problem.
> 
> I'm not the original poster but think I've got the same issue.
> Why does reducing that to a single line solve the problem?
Because otherwise when you call any number and that call ends, it goes
to _X.,2,... because that matches!

-- 
Nabeel Jafferali
tel: 416.491.9136 (toronto)
     646.225.7426 (new york)
fwd: 46990
email/msn : nabeel<at>jafferali.net

Nabeel Jafferali

2005-Jan-04 09:38 UTC

head link

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

> [not-in-service]
> exten => _.,1,Answer
> exten => _.,2,Wait(1)
> exten => _.,3,Playback(the-number-u-dialed)
> exten => _.,4,Playback(not-yet-assigned)
> exten => _.,5,Wait(2)
> exten => _.,6,Hangup
> 
> The above is "included" as the last of several includes, forcing
it to
> be the last choice if no other matches.
Yes, that would be a problem.

Let's imagine the following:

[internal]
include => not-in-service
exten => _2XX,1,Dial(SIP/${EXTEN})

Now, if somebody dials 300, it would go to not-in-service,_.,1 - right?
But if somebody dialed 200, it would go attempt to Dial(SIP/200). If
that call fails or when the call is over, it would go to
not-in-service,_.,2 - since that is a matched next "step" in the
dialplan.
-- 
Nabeel Jafferali
tel: 416.491.9136 (toronto)
     646.225.7426 (new york)
fwd: 46990
email/msn : nabeel<at>jafferali.net

richard Coco

2005-Jan-11 02:58 UTC

head link

[Asterisk-Users] chan_cornet

Hi,
 
we use the oh323 driver (see the post from Joao for installation
http://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg00931.html).
 
in the oh323.conf
[general]
listenPort= 192.x.x.x    /the ip @ of the HG3550
fastStart=yes /*enable fast start

context=voip-h323
codec=G711A
 
in the extensions.conf

exten => _0.,2,Dial,OH323/h323:${EXTEN}ip@HG3550,tr   /* for outgoing to
HiPath

[voip-h323]

include => default /* for incoming from HiPath.

the rest are default settings.

The H4K installation is more difficult. You have to use a HG3550 board.
1. install HG3550 board
2. initial installation of the loadware (use the latest version!!!!!)
3. use WebManagment to configure the routing to Asterisk.
 

marek cervenka <cervajs@fpf.slu.cz> wrote:> i agree... no H.323 support for endpoint (HG3530) but you still have h.323
(for the moment only version 2.0)
> support for ip-trunking (HG3550).
> So what if you have the following setup.
>
>
[OPTIPOINT400_HFA]--[HIPAT4K]----[oh.323]----[ASTERISK]--[OPTIPOINT400_SIP].
how can i configure ip-trunking from HI4K to asterisk?
any example h323 conf for asterisk?

---------------------------------------
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA - http://www.fpf.slu.cz
LCNA - http://lcna.slu.cz
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