Hi,<br><br>I have an interesting problem that the dial out via sip always generates 603 error<br><br>The following is the sip debug<br><br><br>Your help is appreciated.<br><br>CK<br> == Using SIP RTP CoS mark 5<br> -- Executing [998560848@DLPN_DP1:1] Dial("SIP/6100-0000005b", "SIP/13398560848@hkbn2b") in new stack<br>
== Using SIP RTP CoS mark 5<br>Audio is at 113.253.230.26 port 11316<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:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport<br>Max-Forwards: 70<br>From: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as1d554c43<br>
To: <<a href="http://sip:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a>><br>Contact: <<a href="mailto:sip%3A35944101hk@113.253.230.26">sip:35944101hk@113.253.230.26</a>><br>Call-ID: <a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.2.10<br>Date: Sun, 15 Aug 2010 13:47:41 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 2083113394 2083113394 IN IP4 113.253.230.26<br>s=Asterisk PBX 1.6.2.10<br>c=IN IP4 113.253.230.26<br>t=0 0<br>m=audio 11316 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 13398560848@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:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a>><br>
f: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as1d554c43<br>i: <a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>v: SIP/2.0/UDP 113.253.230.26:5060;received=113.253.230.70;rport;branch=z9hG4bK575022bd<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:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a>>;tag=1652716799<br>
f: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as1d554c43<br>i: <a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a><br>
CSeq: 102 INVITE<br>v: SIP/2.0/UDP 113.253.230.26:5060;received=113.253.230.70;rport;branch=z9hG4bK575022bd<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:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport<br>
Max-Forwards: 70<br>From: "cklee@mobile" <<a href="mailto:sip%3A35944101hk@s2hkbntel.net">sip:35944101hk@s2hkbntel.net</a>>;tag=as1d554c43<br>To: <<a href="http://sip:13398560848@s2hkbntel.net:5060">sip:13398560848@s2hkbntel.net:5060</a>>;tag=1652716799<br>
Contact: <<a href="mailto:sip%3A35944101hk@113.253.230.26">sip:35944101hk@113.253.230.26</a>><br>Call-ID: <a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.2.10<br>Content-Length: 0<br><br><br>---<br>Scheduling destruction of SIP dialog '<a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a>' in 6400 ms (Method: INVITE)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br> -- Executing [998560848@DLPN_DP1:2] Hangup("SIP/6100-0000005b", "") in new stack<br> == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on 'SIP/6100-0000005b'<br>
Really destroying SIP dialog '<a href="mailto:34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net">34c9241622c72c7d26b13fdc22d95530@s2hkbntel.net</a>' Method: INVITE<br>ns*CLI> sip set debug off<br>SIP Debugging Disabled<br>
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