[Asterisk-Users] Analog channel stays offhook

[Danial Subhani]

Hi,

We are having a problem with asterisk detecting that an analog ext has been 
put down. This seems only to happen after a number of calls have been made.

We have an FXO port (TDM400P with FXO module) connected to our PBX and are 
using this to test asterisk prior to rolling our for our small office.

What happens is that we make a number of calls to this ext which 1st rings 
a phone (FXS) then after a timeout calls a sip phone. This seems to work 
well but after a while it stops working. What we found is that the analog 
ext seems to be busy ie: it has not hung up.

What we see using 'zap show channel 10' is as follows:

debian*CLI> zap show channel 10
Channel: 10
File Descriptor: 16
Span: 3
Extension:
Dialing: no
Context: from-analog-incoming
Caller ID string:
Destroy: 0
Signalling Type: FXS Kewlstart
Owner: Zap/10-1
Real: Zap/10-1
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Actual Hookstate: Offhook

It seems to remain in this state until we reboot the box.

This is our zapata.conf for the context in question:

[channels]
language=en
context=from-analog-incoming
signalling=fxs_ks
usecallerid=yes
callprogress=no
busydetect=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
channel => 10-11

We have tried it with callprogress=no and callprogress=yes

This is the context from extensions.conf

[from-analog-incoming]
exten => s,1,Dial(${PHONE1}&${PHONE2},10)|tTr
;exten => s,1,Dial(SIP/1337,10)|tTr
exten => s,2,Dial(SIP/dan,10)|tTr
exten => s,3,Congestion
exten => s,4,Hangup
exten => s,102,Voicemail2(b500)
exten => s,103,Congestion
exten => s,104,Hangup
exten => i,1,Hangup
exten => h,1,Hangup


Any ideas ?

Regards
Dan

PRO-NET INTERNET SERVICES LTD