asterisk users - Jul 2007 - Dialtone when automatically picking up.

I'm in the process of setting up a 'phone tree', and are running
into
some problems.  My goal is for users to dial a phone number, the 
asterisk system picks it up, plays the greeting, and users can type 
whatever they want into the system.

What actually happens is users dial the phone number, asterisk picks up 
and additionally goes off-hook on another line, plays the greeting and 
the dialtone over the incoming channel, drops some keypresses (seems to 
only acknowledge the 2nd digit of any length key press), plays 
'invalid', loops back to the beginning (re-plays the message, etc) and 
now behaves normally (no dial tone, accepts all keypresses, etc).  When 
you use a local phone to dial out-in, you can call 3 digit numbers 
(which are dialed 'underneath' asterisk), but if you aren't using a
local
SIP client, then the system does not send the keypresses to the outside 
line.  If the dialplan is to Dial() a number, everything works the way I 
expect.

I essentially copied my dialplan out of the Asterisk: TFOT book, and 
everything seems sane.  I've included the relevant parts of my config, 
and a verbose watch of the call below.  More information is available, 
please let me know.

The hardware is a Sangoma Remora+Shark card, with inbound POTS/PSTN 
analog lines connected to FXO cards.  My testing methodology is both 
using a SIP client to dial the inbound line (therefore going outbound 
to the Phone Company, then back in), and by using a cell phone to dial 
in.

Asterisk 1.2.22 on CentOS 5.0.  kernel 2.6.18-8.el5.  Zaptel drivers are
at 1.4.4, WANPIPE drivers are at 2.3.4-12.


==== configs and debug ===$ cat zapata.conf
;autogenerated by /usr/local/sbin/config-zaptel  do not hand edit
;Zaptel Channels Configurations (zapata.conf)
;
;For detailed zapata options, view /etc/asterisk/zapata.conf.orig

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;Sangoma A200 [slot:6 bus:1 span:1]
context=from-zaptel
group=0
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
signalling = fxs_ks
channel => 4

context=from-zaptel
group=0
signalling = fxs_ks
channel => 5

context=from-zaptel
group=0
signalling = fxs_ks
channel => 6

context=from-zaptel
group=0
signalling = fxs_ks
channel => 7

context=from-zaptel
group=0
signalling = fxs_ks
channel => 8

; this is the fax line, take it out of the regular pool
context=from-fax-line
group=1
signalling = fxs_ks
channel => 9

context=from-zaptel
group=0
signalling = fxs_ks
channel => 10

context=from-zaptel
group=0
signalling = fxs_ks
channel => 11

context=from-zaptel
group=0
signalling = fxs_ks
channel => 12
=== end of zapata.conf ==



$ cat extensions.conf
[globals]
PHESSLER=SIP/phessler
FAXMACHINE=SIP/faxmachine
OUTBOUNDTRUNK=Zap/g0

;; inbound faxes
[from-fax-line]
exten => s,1,Dial(${FAXMACHINE},20)
exten => s,n,Hangup()

;; define a macro to handle everyone the same
;; this way we only have to update one place
;; call it with 'Macro(user,${USERNAME})
[macro-user]
exten => s,1,Dial(${ARG1},20,r)
exten => s,n,Playback(vm-nobodyavail)
exten => s,n,Hangup()

[user-phone-tree]
exten => 305,1,Macro(user,${PHESSLER})

;; from POTS/PSTN
[from-zaptel]
exten => s,1,Answer()
exten => s,n,NoOp(DID is ${DID})
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten(10)

;; let someone from the outside dial our extension directly
include => user-phone-tree

;; invalid entrys
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(from-zaptel,s,1)

exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
=== end of extensions.conf ==


$ asterisk -r
asterisk1*CLI> set verbose 12
Verbosity is at least 12
asterisk1*CLI> 
    -- Starting simple switch on 'Zap/10-1'
    -- Executing Answer("Zap/10-1", "") in new stack
    -- Executing NoOp("Zap/10-1", "DID is ") in new stack
    -- Executing BackGround("Zap/10-1",
"enter-ext-of-person") in new stack
    -- Playing 'enter-ext-of-person' (language 'en')
    -- Executing WaitExten("Zap/10-1", "10") in new stack
;; dialed '305' from the calling phone
    -- Invalid extension '0' in context 'from-zaptel' on
Zap/10-1
  == CDR updated on Zap/10-1
    -- Executing Playback("Zap/10-1", "pbx-invalid") in new
stack
    -- Playing 'pbx-invalid' (language 'en')
    -- Executing Goto("Zap/10-1", "from-zaptel|s|1") in new
stack
    -- Goto (from-zaptel,s,1)
    -- Executing NoOp("Zap/10-1", "DID is ") in new stack
    -- Executing BackGround("Zap/10-1",
"enter-ext-of-person") in new stack
    -- Playing 'enter-ext-of-person' (language 'en')
    -- Executing WaitExten("Zap/10-1", "10") in new stack
    -- Hungup 'Zap/10-1'
asterisk1*CLI>


-- 
"Cleveland?  Yes, I spent a week there one day."

On Wed, 2007-07-25 at 14:05 -0700, Peter Hessler wrote:
>     -- Executing BackGround("Zap/10-1",
"enter-ext-of-person") in new stack
>     -- Playing 'enter-ext-of-person' (language 'en')
>     -- Executing WaitExten("Zap/10-1", "10") in new
stack
> ;; dialed '305' from the calling phone
>     -- Invalid extension '0' in context 'from-zaptel' on
Zap/10-1
You seem to be having DTMF recognition problems here... If I'm reading
this right, Asterisk only saw the 0 when you dialed 305.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.