<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:verdana, helvetica, sans-serif;font-size:10pt"><DIV>I have the following setup:<BR>UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite)<BR><BR>Relevant parts of sip.conf<BR>[general]<BR>port = 5060 ; Port to bind to (SIP is 5060)<BR>bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)<BR>externip = 60.234.100.100 ;External IP address<BR>localnet = 192.168.1.0/255.255.255.0 ;Local network address<BR>allow=all<BR><BR>[1590]<BR>username=1590<BR>type=friend<BR>secret=secret<BR>qualify=no<BR>port=5060<BR>nat=yes<BR><A href="mailto:mailbox=1590@device"><FONT
color=#0000ff>mailbox=1590@device</FONT></A><BR>host=dynamic<BR>dtmfmode=rfc2833<BR>context=test<BR>canreinvite=no<BR>allow=all<BR><BR>[1593]<BR>username=1593<BR>type=friend<BR>secret=secret<BR>qualify=no<BR>port=5060<BR>nat=yes<BR><A href="mailto:mailbox=1593@device"><FONT color=#0000ff>mailbox=1593@device</FONT></A><BR>host=dynamic<BR>dtmfmode=rfc2833<BR>context=test<BR>canreinvite=no<BR>allow=all<BR><BR>I have enabled rtp debugging and notice that Asterisk is receiving no rtp <BR>traffic. When I call from either UA to voicemail for example I see RTP <BR>traffic<BR>e.g. call from 1590<BR>Got RTP packet from 60.234.200.200:38510 (type 0, seq 1245, ts 207620, len <BR>160)<BR>Sent RTP packet to 60.234.200.200:38510 (type 0, seq 61963, ts 34880, len <BR>160)<BR>e.g. call from 1593<BR>Got RTP packet from 60.234.200.200:16470 (type 0, seq 892, ts 316685167, len <BR>240)<BR>Sent RTP packet to 60.234.200.200:16470 (type 0, seq 1156, ts 15360, len <BR>160)<BR><BR>I thought that
with canreinvite=no all audio would go through Asterisk. What <BR>have I missed?</DIV>
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<DIV>Asterisk 1.2.13</DIV>
<DIV>Fedora Core 5<BR><BR>Regards<BR><BR>Cameron <BR></DIV></DIV></div><br>
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