Flavio Miranda
2011-Jul-26 18:20 UTC
[asterisk-users] Scheduling destruction of SIP dialog
Hello, I am receiving the following message all the time, all sip peers, and always finishing with "destructing dialog..". : --- (13 headers 0 lines) --- Sending to 192.168.0.106 : 5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.0.106:5060: OPTIONS sip:2036 at 192.168.0.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK58b8c6b7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk at 192.168.0.254>;tag=as34ab67bd To: <sip:2036 at 192.168.0.106:5060> Contact: <sip:asterisk at 192.168.0.254> Call-ID: 21adef7521218c116309d7784527451c at 192.168.0.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 26 Jul 2011 18:09:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.0.106:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK1228024af6;received=192.168.0.106;rport=5060 From: "Central2" <sip:2036 at 192.168.0.254>;tag=40e337db To: "Central2" <sip:2036 at 192.168.0.254>;tag=as11725d36 Call-ID: 393c15291791541a4628830c0db3acd0 at 192.168.0.106 CSeq: 802 REGISTER Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: <sip:2036 at 192.168.0.106:5060>;expires=60 Date: Tue, 26 Jul 2011 18:09:32 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '393c15291791541a4628830c0db3acd0 at 192.168.0.106' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.0.106:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.254:5060;rport=5060;received=192.168.0.254;branch=z9hG4bK58b8c6b7 From: "asterisk" <sip:asterisk at 192.168.0.254>;tag=as34ab67bd To: <sip:2036 at 192.168.0.106:5060>;tag=0c6ccbbd Call-ID: 21adef7521218c116309d7784527451c at 192.168.0.254 Contact: <sip:2036 at 192.168.0.106:5060> CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS Content-Length: 0 Nay body know what's wrong here ? Thanks! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110726/f9c95b85/attachment.htm>
Kevin P. Fleming
2011-Jul-26 18:26 UTC
[asterisk-users] Scheduling destruction of SIP dialog
On 07/26/2011 02:20 PM, Flavio Miranda wrote:> Hello, > > > I am receiving the following message all the time, all sip peers, and > always finishing with "destructing dialog..". : > > --- (13 headers 0 lines) --- > Sending to 192.168.0.106 : 5060 (no NAT) > Reliably Transmitting (no NAT) to 192.168.0.106:5060: > OPTIONS sip:2036 at 192.168.0.106:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK58b8c6b7;rport > Max-Forwards: 70 > From: "asterisk" <sip:asterisk at 192.168.0.254>;tag=as34ab67bd > To: <sip:2036 at 192.168.0.106:5060> > Contact: <sip:asterisk at 192.168.0.254> > Call-ID: 21adef7521218c116309d7784527451c at 192.168.0.254 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 1.6.2.18 > Date: Tue, 26 Jul 2011 18:09:32 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > > --- > > <--- Transmitting (no NAT) to 192.168.0.106:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.106:5060;branch=z9hG4bK1228024af6;received=192.168.0.106;rport=5060 > From: "Central2" <sip:2036 at 192.168.0.254>;tag=40e337db > To: "Central2" <sip:2036 at 192.168.0.254>;tag=as11725d36 > Call-ID: 393c15291791541a4628830c0db3acd0 at 192.168.0.106 > CSeq: 802 REGISTER > Server: Asterisk PBX 1.6.2.18 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Expires: 60 > Contact: <sip:2036 at 192.168.0.106:5060>;expires=60 > Date: Tue, 26 Jul 2011 18:09:32 GMT > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog > '393c15291791541a4628830c0db3acd0 at 192.168.0.106' in 32000 ms (Method: > REGISTER) > > <--- SIP read from UDP:192.168.0.106:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.254:5060;rport=5060;received=192.168.0.254;branch=z9hG4bK58b8c6b7 > From: "asterisk" <sip:asterisk at 192.168.0.254>;tag=as34ab67bd > To: <sip:2036 at 192.168.0.106:5060>;tag=0c6ccbbd > Call-ID: 21adef7521218c116309d7784527451c at 192.168.0.254 > Contact: <sip:2036 at 192.168.0.106:5060> > CSeq: 102 OPTIONS > Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS > Content-Length: 0 > > Nay body know what's wrong here ?What makes you think something is wrong? Nothing is wrong here, this is perfectly normal. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org