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<P>Here is part of the log from *1, showing a 403 received and 486 passed on (IP addresses, host names and telephone number changed):<BR><BR>--- (10 headers 0 lines)---<BR><-- SIP read from ip.of.asterisk.2:5060:<BR>SIP/2.0 403 Forbidden<BR>Via: SIP/2.0/UDP ip.of.asterisk.1:5060;branch=z9hG4bK596fad3e<BR>From: "1000" <sip:12345@ip.of.asterisk.1>;tag=as46f716d4<BR>To: <sip:01234567890@asterisk.2.com>;tag=as0474d5e5<BR>Call-ID: <A href="mailto:049978b95e32737b4f61c07856176f87@ip.of.asterisk.1" target=_blank>049978b95e32737b4f61c07856176f87@ip.of.asterisk.1</A><BR>CSeq: 103 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Contact: <sip:01234567890@ip.of.asterisk.2><BR>Content-Length: 0<BR><BR>--- (10 headers 0 lines)---<BR>Transmitting (no NAT) to ip.of.asterisk.2:5060:<BR>ACK sip:01234567890@asterisk.2.com SIP/2.0<BR>Via: SIP/2.0/UDP ip.of.asterisk.1:5060;branch=z9hG4bK596fad3e;rport<BR>From: "1000" <sip:12345@ip.of.asterisk.1>;tag=as46f716d4<BR>To: <sip:01234567890@asterisk.2.com>;tag=as0474d5e5<BR>Contact: <sip:12345@ip.of.asterisk.1><BR>Call-ID: <A href="mailto:049978b95e32737b4f61c07856176f87@ip.of.asterisk.1" target=_blank>049978b95e32737b4f61c07856176f87@ip.of.asterisk.1</A><BR>CSeq: 103 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length: 0<BR><BR>---<BR>Feb 1 14:48:12 WARNING[1436]: chan_sip.c:9532 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"1000" <sip:12345@ip.of.asterisk.1>;tag=as46f716d4'<BR> -- SIP/sip_channel-a404 is circuit-busy<BR> == Everyone is busy/congested at this time (1:0/1/0)<BR> -- Executing Busy("SIP/vcs-7b2e", "") in new stack<BR>Transmitting (no NAT) to ip.of.client:5060:<BR>SIP/2.0 486 Busy Here<BR>Via: SIP/2.0/UDP ip.of.client:5060;received=ip.of.client<BR>From: <sip:unknown@0.0.0.0>;tag=d6e314d4-13c4-43e0ca2b-39df1197-4823<BR>To: <sip:01234567890@asterisk.1.com>;tag=as2d609cdc<BR>Call-ID: d6e314d4-13c4-43e0ca2b-39df1197-29-1a214c0<BR>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: <sip:01234567890@ip.of.asterisk.1><BR>Content-Length: 0<BR>X-Asterisk-HangupCause: Call Rejected</P>
<P>The dial plan is a basic:</P>
<P>exten ==> _0Z.,1,Dial(SIP/sip_channel/${EXTEN},30,j)<BR>exten ==> _0Z.,2,Congestion<BR>exten ==> _0Z.,102,Busy</P>
<P>David</P>
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