asterisk users - Jun 2005 - Zap channel not hangingup

Hi,

I am setting up a test call center using *. I am using one Zap channel
(Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip
phones (SjPhone) for call agents. I have setup queues and agents. While
testing I found that if the agent presses * key in soft phone while
attending calls Zap channel gets hung up, and another customer can call.
But if the caller hangs up (for example while announcement is going on)
* does not hangup the call, and the line remains engaged. I have to
restart * to free the line. 

I am attaching all configuration files i have modified and -vvvc output
from *. I am testing this setup inside our pbx network. I plan to use
PSTN as soon as testing is over. I am in India, if that matters. 

I would also appreciate if you can point out any improvements in my
configuration files. I have tried my best to configure this based on the
docs I have read. 

Thanks and Regards,

raj

extensions.conf
---------------

[general]

static=yes
writeprotect=yes

[globals]
[bogon-calls]
exten => _.,1,Congestion

[MainMenu]
exten => s,1,Background(Welcome)
exten => 9,2,Queue(callcenter)
exten => 0,3,Hangup
exten => i,1,Goto,s
exten => t,1,Goto,s

[ivr]
exten => s,1,Answer
exten => s,2,Goto,MainMenu|s|1

[from-sip]
exten => 28,1,AgentLogin(1001)

* output while calling
----------------------

Asterisk Ready.
    -- Starting simple switch on 'Zap/1-1'
Jun  4 12:29:23 NOTICE[2910]: chan_zap.c:5624 ss_thread: Got event 2
(Ring/Answered)...
Jun  4 12:29:27 NOTICE[2910]: chan_zap.c:5624 ss_thread: Got event 2
(Ring/Answered)...
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing Goto("Zap/1-1", "MainMenu|s|1") in new
stack
    -- Goto (MainMenu,s,1)
    -- Executing BackGround("Zap/1-1", "Welcome") in new
stack
    -- Playing 'Welcome' (language 'en')
<Hungup the phone here>
    -- Timeout on Zap/1-1
  == CDR updated on Zap/1-1
    -- Executing Goto("Zap/1-1", "s") in new stack
    -- Goto (MainMenu,t,0)
    -- Timeout on Zap/1-1
  == CDR updated on Zap/1-1
    -- Executing Goto("Zap/1-1", "s") in new stack
    -- Goto (MainMenu,t,0)
Beginning asterisk shutdown....
    -- Hungup 'Zap/1-1'
Executing last minute cleanups
  == Destroying any remaining musiconhold processes
Asterisk cleanly ending (2).

agents.conf
-----------

[agents]
autologoff=15
wrapuptime=5000
recordagentcalls=yes
recordformat=wav
createlink=yes
group=1
ackcall=yes
agent => 1001,1234,Agent1
agent => 1002,1234,Agent2

queues.conf
-----------

[general]

[default]

[callcenter]
monitor-format = wav
announce-holdtime = yes
wrapuptime=15
musiconhold = default
announce = queue-markq
strategy = roundrobin
announce-frequency = 60
context = queue-out
member => Agent/1001
member => Agent/1002

sip.conf
--------

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
context =  from-sip

[mysjphone]
type=friend
host=dynamic
dtmfmode=RFC2833
username=mysjphone
secret=password
context=from-sip
disallow=all
allow=gsm
canreinvite=no
reinvite=no

zapata.conf
-----------
[channels]
signalling=fxs_ks
language=en
context=ivr
channel => 1-4
busydetect=yes

/etc/zaptel.conf
----------------
fxsks = 1-4
loadzone = uk
defaultzone= uk

Hello,

I am following up on a previous mail of the same subject at 
http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html

In a nutshell I have connected my asterisk behind a Siemens HICOM 118E
for a small call center application. The external PSTN calls will land
in HICOM 118E and will get routed to 4 extensions which are connected to
a TDM400P (REV E/F -- 4 FXO modules) I have configured a small IVR in *
which are accessed by calling the said extensions. 

But in this setup when the caller hangs up Zap channel is not detecting
it and goes to time out. A sample output is given at the end of the
mail. I am also having echo problems. 

Do I have to make any additional settings to get this working? All my
configurations are available at
http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html

I have been trying to get this working for quite some time and any help
will be much appreciated.

regards,

raj

as a suggestion, please play a little with the next parameters in
zapata.conf read the docs in voip-info about these parameters an may
me you will be able to fix your problem.

echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=-4
immediate=yes
busydetect=yes
callprogress=yes

i think the values that you have to play with more are rxgain, txgain,
callprogress and immediate

best regards.
On 7/16/05, rajkumars@asianetindia.com <rajkumars@asianetindia.com>
wrote:
> Hello,
> 
> I am following up on a previous mail of the same subject at
> http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html
> 
> In a nutshell I have connected my asterisk behind a Siemens HICOM 118E
> for a small call center application. The external PSTN calls will land
> in HICOM 118E and will get routed to 4 extensions which are connected to
> a TDM400P (REV E/F -- 4 FXO modules) I have configured a small IVR in *
> which are accessed by calling the said extensions.
> 
> But in this setup when the caller hangs up Zap channel is not detecting
> it and goes to time out. A sample output is given at the end of the
> mail. I am also having echo problems.
> 
> Do I have to make any additional settings to get this working? All my
> configurations are available at
> http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html
> 
> I have been trying to get this working for quite some time and any help
> will be much appreciated.
> 
> regards,
> 
> raj
> 
> 
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