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<DIV><FONT face=Arial size=2>I am running Asterisk 1.0.9 and have been running
all my calls through a VSP over a IAX2 trunk however we have recently
purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through
Optus. I can make outgoing calls via it fine, however incoming calls
are dropped after a few seconds ( or as soon as a command like Playback, or the
call is picked up if forwarded to a SIP extensions ). </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have upgraded libpri and zaptel to trunk, but I
don't want to upgrade Asterisk to 1.2 until I've got this all sorted, one
problem at a time!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here are my configs :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>/etc/zaptel.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2># Global data</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>span=1,1,0,ccs,hdb3,crc4<BR>bchan=1-15,17<BR>dchan=16<BR>unused=18-31</FONT></DIV>
<DIV><FONT face=Arial size=2>loadzone = au<BR>defaultzone = au</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>/etc/asterisk/zapata.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[channels]<BR>context=from-onramp</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>overlapdial=yes<BR>priindication =
outofband<BR>switchtype=euroisdn<BR>pridialplan=unknown<BR>prilocaldialplan=unknown<BR>signalling=pri_cpe</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>echocancelwhenbridged=yes<BR>echocancel=128<BR>echotraining=800</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>rxgain=5 ; 0<BR>txgain=-4.5 ;
0<BR>busydetect=no<BR>pridialplan=local<BR>internationalprefix=0011<BR>nationalprefix=0<BR>usecallerid=yes<BR>hidecallerid=no<BR>callprogress=no</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>group=0<BR>channel => 1-15,17<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>/etc/asterisk/extensions:</FONT></DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT>
<DIV><FONT face=Arial size=2>[from-onramp]<BR>;exten =>
s,1,Playback(custom/aa_1)<BR>exten => s,1,Dial(SIP/116)<BR>exten =>
h,1,Hangup<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>and here's some log info:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>asterisk*CLI> pri intense debug span
1<BR>Enabled EXTENSIVE debugging on span
1<BR>asterisk*CLI><BR>asterisk*CLI><BR>asterisk*CLI><BR>&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
-- Going to extension s|1 because of Complete received<BR> --
Executing Dial("Zap/1-1", "SIP/116") in new stack<BR> --
Called 116<BR> -- Accepting call from '' to 's' on channel
0/1, span 1<BR> -- SIP/116-5a95 is
ringing<BR>&&&&&&&&&&&&&&&&&
-- SIP/116-5a95 answered Zap/1-1<BR> == Spawn extension (from-onramp, s,
1) exited non-zero on 'Zap/1-1'<BR> -- Executing
Hangup("Zap/1-1", "") in new stack<BR> == Spawn extension (from-onramp, h,
1) exited non-zero on 'Zap/1-1'<BR> -- Hungup
'Zap/1-1'<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>and going straight to a Playback command rather
than SIP extension:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>asterisk*CLI> pri intense debug span
1<BR>Enabled EXTENSIVE debugging on span
1<BR>%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-- Going to extension s|1 because of Complete received<BR> --
Executing Answer("Zap/2-1", "") in new stack<BR> -- Accepting
call from '' to 's' on channel 0/2, span 1<BR> == Spawn extension
(from-onramp, s, 1) exited non-zero on 'Zap/2-1'<BR> --
Executing Hangup("Zap/2-1", "") in new stack<BR> == Spawn extension
(from-onramp, h, 1) exited non-zero on 'Zap/2-1'<BR> -- Hungup
'Zap/2-1'<BR><BR></DIV></FONT>
<DIV><FONT face=Arial size=2> </DIV>
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