<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1264" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I've got Asterisk configured and working (sort of)
with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX).
This * box is on a 'live', non-nat IP address.</FONT></DIV>
<DIV><FONT face=Arial size=2>I also have a couple of budgetone phones, one
behind NAT and one not. When I place an outgoing call, I get the following
messages:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-- Executing Dial("SIP/filbert-9876",
"CAPI/288:333") in new stack<BR> -- creating pipe for
PLCI=-1<BR> > sent CONNECT_REQ MN
=0x5<BR> -- Called 288:333<BR> -- Setting up
echo canceller (PLCI=0x201, function=1, options=2,
tail=64)<BR> > sent FACILITY_REQ
(PLCI=0x201)<BR> -- CAPI[contr1/288]/0 answered
SIP/filbert-9876<BR> -- Echo canceller successfully set up
(PLCI=0x201)<BR>WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): Maximum
retries exceeded on call <A
href="mailto:03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46">03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46</A>
for seqno 102 (Request)<BR> -- CAPI
Hangingup<BR> > sent DISCONNECT_B3_REQ
NCCI=0xa0201<BR> > sent DISCONNECT_REQ
PLCI=0x201<BR> -- removed pipe for PLCI = 0x201<BR> ==
Spawn extension (sip, 9333, 1) exited non-zero on
'SIP/filbert-9876'<BR>WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt):
Maximum retries exceeded on call <A
href="mailto:03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46">03ed3583-2cba-cbfc-a86f-605f7f57f024@213.152.57.46</A>
for seqno 103 (Request)<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>I can hear the voicemail service (extn. 333) answer
correctly, but then after about 5 seconds i'll get the WARNING message and the
system will hangup.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Here's a snippet from my sip.conf
file:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-------</FONT></DIV>
<DIV><FONT face=Arial size=2>[general]<BR>port = 5060<BR>bindaddr =
0.0.0.0</FONT></DIV>
<DIV><FONT face=Arial size=2>context =
sip-incoming<BR>srvlookup=no<BR>qualify=yes<BR>disallow=all<BR>allow=alaw<BR>allow=ulaw</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[filbert]<BR>type=friend<BR>host=dynamic<BR>dtmfmode=info<BR>context=sip<BR>callerid="Jon
Fautley"
<200><BR>nat=yes<BR>pickupgroup=1<BR>reinvite=no<BR>canreinvite=no<BR>disallow=all<BR>allow=ulaw<BR>--------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any ideas?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Many thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Jon</DIV></FONT></BODY></HTML>