Hi, I have an interesting problem that the dial out via sip always generates 603 error
The following is the sip debug Your help is appreciated. CK == Using SIP RTP CoS mark 5 -- Executing [998560...@dlpn_dp1:1] Dial("SIP/6100-0000005b", "SIP/13398560...@hkbn2b") in new stack == Using SIP RTP CoS mark 5 Audio is at 113.253.230.26 port 11316 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.80.89.139:5060: INVITE sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: "ck...@mobile" <sip:3594410...@s2hkbntel.net<sip%3a3594410...@s2hkbntel.net> >;tag=as1d554c43 To: <sip:13398560...@s2hkbntel.net:5060> Contact: <sip:3594410...@113.253.230.26 <sip%3a3594410...@113.253.230.26>> Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.10 Date: Sun, 15 Aug 2010 13:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 2083113394 2083113394 IN IP4 113.253.230.26 s=Asterisk PBX 1.6.2.10 c=IN IP4 113.253.230.26 t=0 0 m=audio 11316 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 13398560...@hkbn2b <--- SIP read from UDP:203.80.89.139:5060 ---> SIP/2.0 100 Trying t: <sip:13398560...@s2hkbntel.net:5060> f: "ck...@mobile" <sip:3594410...@s2hkbntel.net<sip%3a3594410...@s2hkbntel.net> >;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:203.80.89.139:5060 ---> SIP/2.0 487 Request Terminated t: <sip:13398560...@s2hkbntel.net:5060>;tag=1652716799 f: "ck...@mobile" <sip:3594410...@s2hkbntel.net<sip%3a3594410...@s2hkbntel.net> >;tag=as1d554c43 i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.230.26:5060 ;received=113.253.230.70;rport;branch=z9hG4bK575022bd k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 203.80.89.139:5060: ACK sip:13398560...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport Max-Forwards: 70 From: "ck...@mobile" <sip:3594410...@s2hkbntel.net<sip%3a3594410...@s2hkbntel.net> >;tag=as1d554c43 To: <sip:13398560...@s2hkbntel.net:5060>;tag=1652716799 Contact: <sip:3594410...@113.253.230.26 <sip%3a3594410...@113.253.230.26>> Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.10 Content-Length: 0 --- Scheduling destruction of SIP dialog ' 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [998560...@dlpn_dp1:2] Hangup("SIP/6100-0000005b", "") in new stack == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on 'SIP/6100-0000005b' Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' Method: INVITE ns*CLI> sip set debug off SIP Debugging Disabled
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