Here is the sip log<br><br>ns*CLI> sip set debug peer hkbn2b<br>SIP Debugging Enabled for IP: <a href="http://203.80.89.139:5060">203.80.89.139:5060</a><br>[Oct 15 06:35:19] NOTICE[2462]: chan_sip.c:18334 handle_response_register: Outbound Registration: Expiry for <a href="http://sip.voipuser.org">sip.voipuser.org</a> is 120 sec (Scheduling reregistration in 105 s)<br>
== Using SIP RTP TOS bits 184<br> == Using SIP RTP CoS mark 5<br> -- Executing [8935944101@DLPN_DP1:1] Dial("SIP/6100-00000006", "SIP/35944101@hkbn2b,,r") in new stack<br> == Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>Audio is at 113.253.226.153 port 10650<br>Adding codec 0x8 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to <a href="http://203.80.89.139:5060">203.80.89.139:5060</a>:<br>
INVITE <a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport<br>Max-Forwards: 70<br>From: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as12eb85f9<br>
To: <<a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a>><br>Contact: <<a href="mailto:sip%3A35944101hk@113.253.226.153">sip:35944101hk@113.253.226.153</a>><br>Call-ID: <a href="mailto:3f603bea2560e9b836ea250932486935@s2hkbntel.net">3f603bea2560e9b836ea250932486935@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.2.12<br>Date: Thu, 14 Oct 2010 22:35:23 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 241<br><br>v=0<br>o=root 316173620 316173620 IN IP4 113.253.226.153<br>s=Asterisk PBX 1.6.2.12<br>c=IN IP4 113.253.226.153<br>t=0 0<br>m=audio 10650 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called 35944101@hkbn2b<br><br><--- SIP read from UDP:<a href="http://203.80.89.139:5060">203.80.89.139:5060</a> ---><br>SIP/2.0 100 Trying<br>t: <<a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a>><br>
f: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as12eb85f9<br>i: <a href="mailto:3f603bea2560e9b836ea250932486935@s2hkbntel.net">3f603bea2560e9b836ea250932486935@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>v: SIP/2.0/UDP 113.253.226.153:5060;received=113.253.226.174;rport;branch=z9hG4bK1880eaca<br>Server: MCS5x00_3.0<br>k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec<br>l: 0<br><br><br><-------------><br>
--- (9 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://203.80.89.139:5060">203.80.89.139:5060</a> ---><br>SIP/2.0 487 Request Terminated<br>t: <<a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a>>;tag=781480306<br>
f: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as12eb85f9<br>i: <a href="mailto:3f603bea2560e9b836ea250932486935@s2hkbntel.net">3f603bea2560e9b836ea250932486935@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>v: SIP/2.0/UDP 113.253.226.153:5060;received=113.253.226.174;rport;branch=z9hG4bK1880eaca<br>k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec<br>l: 0<br><br><br><-------------><br>--- (8 headers 0 lines) ---<br>
Transmitting (NAT) to <a href="http://203.80.89.139:5060">203.80.89.139:5060</a>:<br>ACK <a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport<br>
Max-Forwards: 70<br>From: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as12eb85f9<br>To: <<a href="http://sip:35944101@s2hkbntel.net:5060">sip:35944101@s2hkbntel.net:5060</a>>;tag=781480306<br>
Contact: <<a href="mailto:sip%3A35944101hk@113.253.226.153">sip:35944101hk@113.253.226.153</a>><br>Call-ID: <a href="mailto:3f603bea2560e9b836ea250932486935@s2hkbntel.net">3f603bea2560e9b836ea250932486935@s2hkbntel.net</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.2.12<br>Content-Length: 0<br><br><br>---<br>Scheduling destruction of SIP dialog '<a href="mailto:3f603bea2560e9b836ea250932486935@s2hkbntel.net">3f603bea2560e9b836ea250932486935@s2hkbntel.net</a>' in 6400 ms (Method: INVITE)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br> -- Executing [8935944101@DLPN_DP1:2] Hangup("SIP/6100-00000006", "") in new stack<br> == Spawn extension (DLPN_DP1, 8935944101, 2) exited non-zero on 'SIP/6100-00000006'<br>
[Oct 15 06:35:23] NOTICE[2462]: chan_sip.c:11601 sip_reregister: -- Re-registration for <a href="mailto:8887109919@sip.pennytel.com">8887109919@sip.pennytel.com</a><br>Reliably Transmitting (NAT) to <a href="http://203.80.89.139:5060">203.80.89.139:5060</a>:<br>
OPTIONS sip:<a href="http://s2hkbntel.net">s2hkbntel.net</a> SIP/2.0<br>Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK703ea06a;rport<br>Max-Forwards: 70<br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@sip.etransmed.net">sip:asterisk@sip.etransmed.net</a>>;tag=as1d0ccbd8<br>
To: <sip:<a href="http://s2hkbntel.net">s2hkbntel.net</a>><br>Contact: <<a href="mailto:sip%3Aasterisk@113.253.226.153">sip:asterisk@113.253.226.153</a>><br>Call-ID: <a href="mailto:67f6129e02db3377276c62f209913543@sip.etransmed.net">67f6129e02db3377276c62f209913543@sip.etransmed.net</a><br>
CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX 1.6.2.12<br>Date: Thu, 14 Oct 2010 22:35:23 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Length: 0<br>
<br><br>---<br><br><--- SIP read from UDP:<a href="http://203.80.89.139:5060">203.80.89.139:5060</a> ---><br>SIP/2.0 100 Trying<br>t: <sip:<a href="http://s2hkbntel.net">s2hkbntel.net</a>><br>f: "asterisk" <<a href="mailto:sip%3Aasterisk@sip.etransmed.net">sip:asterisk@sip.etransmed.net</a>>;tag=as1d0ccbd8<br>
i: <a href="mailto:67f6129e02db3377276c62f209913543@sip.etransmed.net">67f6129e02db3377276c62f209913543@sip.etransmed.net</a><br>CSeq: 102 OPTIONS<br>v: SIP/2.0/UDP 113.253.226.153:5060;received=113.253.226.174;rport;branch=z9hG4bK703ea06a<br>
Server: MCS5x00_3.0<br>k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec<br>l: 0<br><br><br><-------------><br>--- (9 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://203.80.89.139:5060">203.80.89.139:5060</a> ---><br>
SIP/2.0 404 Not Found<br>t: <sip:<a href="http://s2hkbntel.net">s2hkbntel.net</a>>;tag=820879923<br>f: "asterisk" <<a href="mailto:sip%3Aasterisk@sip.etransmed.net">sip:asterisk@sip.etransmed.net</a>>;tag=as1d0ccbd8<br>
i: <a href="mailto:67f6129e02db3377276c62f209913543@sip.etransmed.net">67f6129e02db3377276c62f209913543@sip.etransmed.net</a><br>CSeq: 102 OPTIONS<br>v: SIP/2.0/UDP 113.253.226.153:5060;received=113.253.226.174;rport;branch=z9hG4bK703ea06a<br>
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec<br>l: 0<br><br><br><-------------><br>--- (8 headers 0 lines) ---<br>Really destroying SIP dialog '<a href="mailto:67f6129e02db3377276c62f209913543@sip.etransmed.net">67f6129e02db3377276c62f209913543@sip.etransmed.net</a>' Method: OPTIONS<br>
<br><br><br><div class="gmail_quote">On Thu, Oct 14, 2010 at 7:55 AM, Paul Belanger <span dir="ltr"><<a href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im">On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk <<a href="mailto:asterisk@ck-lee.com">asterisk@ck-lee.com</a>> wrote:<br>
> Appreciate if help or direction can be provided.<br>
><br>
</div>21.6.2 603 Decline<br>
<br>
The callee's machine was successfully contacted but the user<br>
explicitly does not wish to or cannot participate. The response MAY<br>
indicate a better time to call in the Retry-After header field. This<br>
status response is returned only if the client knows that no other<br>
end point will answer the request.<br>
<br>
<a href="http://www.ietf.org/rfc/rfc3261.txt" target="_blank">http://www.ietf.org/rfc/rfc3261.txt</a><br>
<br>
Collect a SIP trace and see if a reason is supplied.<br>
<br>
--<br>
Paul Belanger | dCAP<br>
Polybeacon | Consultant<br>
Jabber: <a href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a> | IRC: pabelanger (Freenode)<br>
<a href="http://blog.polybeacon.com" target="_blank">blog.polybeacon.com</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>