[Asterisk-Users] Re: 'I'nvalid extension handling problems,
even with workaround
telmo at n1.com
telmo at n1.com
Wed Dec 22 20:27:43 MST 2004
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 at 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 at 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 at 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 at 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 at 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 at 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 at voiptalk
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' Request 102: Found
Acked pending invite 102
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 102: Found
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 102: Not Found
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' Request 103: Found
Acked pending invite 103
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 103: Found
build_route: Record-Route hop: <sip:902 at 82.145.32.73;ftag=as3688ce90;lr=on>
build_route: Contact hop: <sip:902 at 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 at 192.168.1.1;user=phone>' failed for '192.168.1.201'
Auto destroying call 'd761b1512a710969 at 192.168.1.201'
Scheduled a timeout # 55
Stopping retransmission on '327b23c6643c98696633487374b0dc51 at 192.168.1.1' of
Request 107: Found
Stopping retransmission on '327b23c6643c98696633487374b0dc51 at 192.168.1.1' of
Request 108: Found
Registration successful
Cancelling timeout 55
Auto destroying call '8a7ae46c9e0d1370 at 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 at 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.
----
SeeqMail - the only email you'll ever need Starts Here
Sign up for FREE personalized email today: http://www.seeqmail.com
http://www.Grassroots.org/ - Make Change!
----
More information about the asterisk-users
mailing list