wendell hamilton wrote:
Please excuse the top-posting.
... so we are faster at the solution, ... ;-)
In features.conf, uncomment transferdigittimeout and adjust its timing as
desired. You may also want to uncomment and adjust featuredigittimeout to a
higher value as well.
That was it!!! Now it works!!!
Also, since the dialplan does first match, you can eliminate the problem by
putting the 4 digit extensions before the 3 digit extensions in the dialplan.
See the "match as you go" section at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
Thank you for the link, btw. your comment above does not "match" the
link. Copy of the important part of your provided link:
Example
FooBar Incorporated wants their incoming telephone calls to be
answered with a voice message welcoming the caller and inviting them
to choose which extension they want. FooBar has six telephone
extensions. Their extension numbers are 1, 2, 21, 22, 31, 32. So this
is the context created for incoming calls for FooBar Incorporated:
[incoming]
exten => s,1,Background(welcome-to-foobar-incorporated)
exten => 1,1,Dial(Zap/1)
exten => 2,1,Dial(Zap/2)
exten => 21,1,Dial(Zap/3)
exten => 22,1,Dial(Zap/4
exten => 31,1,Dial(Zap/5)
exten => 32,1,Dial(Zap/6)
When you call FooBar, Asterisk plays the
"welcome-to-foobar-incorporated.gsm" sound file. After that, having
run out of commands to execute, it waits for you to dial something.
This is what Asterisk would do if you dialed various options:
Number Dialed Asterisk's Action
1 Immediately performs Dial (Zap/1)
2 Waits for timeout, then performs Dial(Zap/2)
21 Immediately performs Dial (Zap/3)
22 Immediately performs Dial (Zap/4)
3 Waits for timeout, then hangs up.
31 Immediately performs Dial (Zap/5)
32 Immediately performs Dial (Zap/6)
4 Immediately hangs up.
Note that when a caller tries to dial extension 2, they are not
connected immediately. Asterisk waits to see if the caller dials more
digits, to determine whether the caller wants extension 2 or 21 or 22.
As callers would like to be connected immediately if possible, it
would be more user-friendly to avoid using ambiguous extension numbers.
Thanks for the solution, ....
bye
Ronald
HTH
routerguy
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits
David Gagnon wrote:
Ronald,
Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.
I'm sure most of the people here don't understand how you try to
transfer.
David
David,
I am not sure how the explanation how to punch the keys changes
something, .... ;-.)
Ok, here we go:
Snom:
pick up the phone and hit ##6014 followed by [ok]
Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end
# needed, since the phone 601 starts to ring as soon I reach 1.
In my opinion Asterisk remembers all numbers and therefore it does not
wait for the 4, since it found a match. This is in VoIP (in my opinion)
wrong, since overlapping numbers are allowed.
Sip message / dtmf, this is something different! How is the transfer
made? Maybe snom does send a sip message, while the noname only send
dtmf tones.
bye
Ronald
-----Message d'origine-----
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits
Koopmann, Jan-Peter wrote:
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?
Because the transfer button on the SNOM is using a totally different
mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit "OK". At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question,
than it would be for each dialing, but it isn't.
You mentioned SIP message and that makes me wonder! Are we not using
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of "tones", than why is it different if it is in a
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on
the system to take a longer break between "tones" still as a string?
Back to the dialplan:
A Voip number can have different length of digits. Each number is seen
as a complete "picture", and so a three digit and a four digit number is
something different. While in the legacy telephony the digits are worked
down one by one and if there is no more use of the digits, they are just
garbage and will be not used. Unlike in VoIP, where you can have a three
digit number and if you dial four digit, than it is a WRONG number !!!!
I just verified that: I dialed from 601 to 61522, however, 61522 does
not exist, but 615 exists. Guess what? I get a busy tone! That should
proof my thoughts (and that without yelling, ... hehehehe)
bye
Ronald
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