[Asterisk-Users] Bristuffed Asterisk: Hangup problems
Tim Robinson
timweb at txrx.org.uk
Thu May 11 02:17:48 MST 2006
There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines. This might be causing confusion?
Here is my config for a BT ISDN2e line here in UK. I don't think I have
the problems you report.
;TE mode - for ISDN line
nocid=Unavailable
withheldcid=Withheld
Language=en
usecallerid=yes
pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=0
internationalprefix=00
switchtype = euroisdn
signalling = bri_cpe_ptmp
echocancel=yes
echocancelwhenbridged=no
immediate=no
overlapdial=yes
group = 1
context=isdn-in
callgroup=1
channel => 1-2
Rgds
Tim
Jeroen Zwarts wrote:
>Hello,
>
>I have a problem with the Bristuffed version of Asterisk. I have tried
>Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have
>the same problem it seems:
>
>The setup:
>
>A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15.
>Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly.
>Grandstream gxp-2000 as a SIP phone, and a normal mobile phoneaccount to
>test the ISDN connection to the outside world.
>
>
>I try to upgrade the above setup to a newer version of Bristuff (to test if
>the CDR problem is solved) so I download a new bristuff and install it. The
>installation seems to go OK, the zaptel and zaphfc modules load with no
>problems at all. But when I hang up a call, the line is not totally
>disconnected. For instance: When I call from my internal SIP phone to the
>mobile, and I hangup the SIP phone while the other side hasn't picked up,
>the mobile phone keeps on ringing. Or if I call from the outside to the SIP
>phone, and hang up the mobile phone after a conversation, the SIP connection
>to Asterisk is still there. Or the other way around: When I have a
>conversation from the SIP phone to the mobile phone, and I hang up the SIP
>phone, the connection to the mobile phone is still there.
>
>I have tried a clean install of Asterisk+Bristuff as well as an upgrade from
>the working Asterisk 1.2.0, but it gives me the same problem.
>
>The only really strange thing I find in the logs that might have to do
>something with this is the following line on the verbosed console:
>chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1
>
>
>my zaptel.conf:
>---------------------
>loadzone = nl
>defaultzone=nl
>
>span=1,1,3,ccs,ami
>bchan=1-2
>dchan=3
>-------------------
>
>
>and my zapata.conf:
>-------------------
>[channels]
>language=nl
>context=inbound
>switchtype=euroisdn
>pridialplan=dynamic
>prilocaldialplan=local
>internationalprefix = 00
>nationalprefix = 0
>signalling=bri_cpe_ptmp
>
>rxwink=300
>usecallerid=yes
>cidsignalling=dtmf
>cidstart=ring
>hidecallerid=no
>callwaiting=yes
>usecallingpres=yes
>callwaitingcallerid=yes
>threewaycalling=yes
>transfer=yes
>cancallforward=yes
>callreturn=yes
>
>echocancel=yes
>echocancelwhenbridged=yes
>echotraining=yes
>rxgain=4.0
>txgain=0.0
>
>group=1
>callgroup=1
>pickupgroup=1
>immediate=no
>
>callerid=asreceived
>busydetect=yes
>busycount=8
>busypattern=500,500
>
>channel => 1-2
>-------------------
>
>
>I also made a full log and a console log of this problem. They can be found
>at:
>
>http://www.borndgtl.cistron.nl/consolelog.txt
>
>http://www.borndgtl.cistron.nl/fulllog.txt
>
>
>
>Anyone has an idea where to look for a solution for this problem?
>
>Thanks!
>
>Jeroen Zwarts
>Born Digital
>the Netherlands
>
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