[asterisk-users] 603 error
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/6100-005b, 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.netsip%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.netsip%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.netsip%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.netsip%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/6100-005b, ) in new stack == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on 'SIP/6100-005b' Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' Method: INVITE ns*CLI sip set debug off SIP Debugging Disabled -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 603 Error
- Original Message - From: Olle E Johansson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 03, 2007 8:59 AM Subject: Re: [asterisk-users] 603 Error 2 apr 2007 kl. 10.16 skrev Dovid B: Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP XXX.XXX.XX.XXX: 5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: sip:XXX.XXX.XX.XX;tag=a21dc3d8dd92817bo0 To: sip:XXX.XXX.XX.XX;tag=as7b187bff Call-ID: [EMAIL PROTECTED] CSeq: 112226 NOTIFY User-Agent: Blah Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Your server is sending a NOTIFY that the ITSP's server doesn't like. Propably a mailbox notification. Not a critical error, just a configuration issue. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! Thanks. Our lines were down so I was just guessing. Ended up being the ITSP's fault. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 603 Error
2 apr 2007 kl. 10.16 skrev Dovid B: Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP XXX.XXX.XX.XXX: 5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: sip:XXX.XXX.XX.XX;tag=a21dc3d8dd92817bo0 To: sip:XXX.XXX.XX.XX;tag=as7b187bff Call-ID: [EMAIL PROTECTED] CSeq: 112226 NOTIFY User-Agent: Blah Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Your server is sending a NOTIFY that the ITSP's server doesn't like. Propably a mailbox notification. Not a critical error, just a configuration issue. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 603 Error
Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP XXX.XXX.XX.XXX:5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: sip:XXX.XXX.XX.XX;tag=a21dc3d8dd92817bo0 To: sip:XXX.XXX.XX.XX;tag=as7b187bff Call-ID: [EMAIL PROTECTED] CSeq: 112226 NOTIFY User-Agent: Blah Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
