[Asterisk-Dev] Nightmare on disconnecting Zap and SIP channel
Asterisk Mania
astmania at gmail.com
Thu Oct 28 00:39:58 MST 2004
Hi all,
I have a nightmare working on disconnecting Zap and SIP channels. I've
been battling for almost 3 days with no avail.
My setup is something like this:
[GSM PHONE] --> [GSM Mobile Trunk Gateway] --> [TDM04B] --> [Asterisk]
--> [International VOIP provider]
I called from a GSM mobile phone to GSM trunk gateway then connected
to FXO and Asterisk for outbound calls. When I end up the call, the
Zap and SIP channel does not disconnect and the channels are still
active. So soft hangup is the solution to destroy the active channels
but sometimes the Zap channel is unusable and need to reload the
zaptel and wcfxs driver and restart asterisk to make the zap channel
work. I tried to check my GSM trunk gateway, using a voltage meter
just to know if it is sending a disconnect tone or changing the
voltage and it does. So it seems that Zaptel does not know how to deal
with it and the channel are still active. I would like to ask what are
the description of the Zaptel card, like loop current detection,
polarity reversal, disconnect tones etc etc and how does it deal with
that.
I also tried to use different release of zaptel drivers, from old cvs
to fresh cvs and stable release of zaptel driver from Digium but no
luck. also played some settings on zapata.conf.
my zapata.conf
[channels]
context=gsmmenu
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;echocancel=256
;echotraining=800
callwaiting=no
busydetect=1
busycount=7
relaxdtmf=yes
rxgain=-1.0
txgain=1.0
immediate=no
callprogress=yes
musiconhold=default
usecallerid=no
channel => 1-8
my zaptel.conf
fxsks=1-4
loadzone = us
defaultzone=us
my extensions.conf
[gsmmenu]
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,Background(agent-pass)
exten => s,4,Authenticate(/etc/asterisk/pincode,a)
exten => s,5,Wait,2
exten => s,6,DigitTimeout,5
exten => s,7,ResponseTimeout,10
exten => s,8,Background(gsmmenu)
exten => _1NXXNXXXXXX,1,Dial(SIP/vpprovider/${EXTEN})
exten => _1NXXNXXXXXX,2,Hangup
exten => i,1,Playback,invalid
exten => i,2,Goto(s,6)
exten => t,1,Goto(s,6)
Hope anyone can help me.
Best regards,
Asterisk Mania
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