What is your provider seeing? Many providers send re-INVITEs at 15 minutes. Many firewalls have closed their port before this due to UDP timeouts. I have a whitepaper that I wrote on this subject; I will see if I can dig it up.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Florian Wolters Sent: Thursday, March 21, 2013 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes Hello, > I solved it by moving Asterisk 1.6 to Asterisk 1.4. > > Try asterisk 1.4 or 1.8 on a test box and see how it goes. I did try the latest 1.8.2x release already without any improvement. Currently running is a Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 as the tcpdump says (little mistake to my last mail). I also played around with "canreinvite". But regardless of the setting (yes/no) I still get disconnects after 15 minutes. I just tried to accept session-timers, but this has no connection to this issue either. So I turned on SIP debug for this host and analyszed it with wireshark. The last packets show an INVITE from my provider, that is answered by my Asterisk with "200 OK, with session description". What follows is an ACK by the provider and immediately a BYE sent by the provider. So for me it looks like the provider is disconnecting the call. I could not see any reason or hangup cause for this in the dump. Are there error messages for this that can be seen in the protocol? The tcpdump (the last few packets) shows: --- 8< snip --- 13:37:28.258566 IP (tos 0x0, ttl 64, id 44187, offset 0, flags [DF], proto TCP (6), length 611) 172.16.0.2.44929 > 217.0.17.170.5060: Flags [P.], cksum 0xf764 (incorrect -> 0xd1be), seq 4568:5139, ack 4057, win 45600, length 571 13:37:28.277390 IP (tos 0xc0, ttl 55, id 4807, offset 0, flags [DF], proto TCP (6), length 547) 217.0.17.170.5060 > 172.16.0.2.44929: Flags [P.], cksum 0x2c63 (correct), seq 4057:4564, ack 5139, win 65535, length 507 13:37:28.277415 IP (tos 0x0, ttl 64, id 44188, offset 0, flags [DF], proto TCP (6), length 40) 172.16.0.2.44929 > 217.0.17.170.5060: Flags [.], cksum 0xf529 (incorrect -> 0xdc6d), ack 4564, win 45600, length 0 13:37:54.240304 IP (tos 0xc0, ttl 25, id 14090, offset 0, flags [none], proto UDP (17), length 1255) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1227 INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 Max-Forwards: 70 To: <sip:[email protected]:5060>;tag=as77f2fb84 From: <sip:[email protected];user=phone>;tag=8f233b97 Call-ID: [email protected] Contact: <sip:[email protected]:5083;transport=tcp>;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel" CSeq: 1939619 INVITE Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Disposition: session Content-Length: 297 v=0 o=- 558131575 1701401067 IN IP4 217.0.17.170 s=Phone Call via hiQ9200 SIPCA c=IN IP4 217.0.1.67 t=0 0 m=audio 16884 RTP/AVP 8 100 b=AS:110 b=RS:1375 b=RR:4125 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sqn: 0 a=sendrecv a=ptime:20 13:37:54.240497 IP (tos 0xc0, ttl 25, id 14091, offset 0, flags [none], proto UDP (17), length 1222) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1194 INVITE sip:[email protected]:5060;transport=TCP SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK86e8a99ff6e15f3e72053880758bf877.9856a4e1 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709 Max-Forwards: 70 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd From: <sip:[email protected]>;tag=f18b4044 Call-ID: [email protected] Contact: <sip:[email protected]:5082;transport=tcp>;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel" CSeq: 1939639 INVITE Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Disposition: session Content-Length: 224 v=0 o=- 1028575251 1704720679 IN IP4 217.0.17.170 s=Basic Session c=IN IP4 217.0.1.81 t=0 0 m=audio 17120 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 13:37:54.240593 IP (tos 0x0, ttl 64, id 43415, offset 0, flags [none], proto UDP (17), length 782) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 754 SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b;received=217.0.17.170;rport=5060 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 From: <sip:[email protected];user=phone>;tag=8f233b97 To: <sip:[email protected]:5060>;tag=as77f2fb84 Call-ID: [email protected] CSeq: 1939619 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Length: 0 13:37:54.240752 IP (tos 0x0, ttl 64, id 43416, offset 0, flags [none], proto UDP (17), length 1064) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 1036 SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b;received=217.0.17.170;rport=5060 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 From: <sip:[email protected];user=phone>;tag=8f233b97 To: <sip:[email protected]:5060>;tag=as77f2fb84 Call-ID: [email protected] CSeq: 1939619 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Type: application/sdp Content-Length: 253 v=0 o=root 515584563 515584563 IN IP4 79.253.136.104 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 79.253.136.104 t=0 0 m=audio 16240 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=ptime:20 a=sendrecv 13:37:54.240976 IP (tos 0x0, ttl 64, id 43417, offset 0, flags [none], proto UDP (17), length 813) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 785 SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK86e8a99ff6e15f3e72053880758bf877.9856a4e1;received=217.0.17.170;rport=5060 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709 From: <sip:[email protected]>;tag=f18b4044 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd Call-ID: [email protected] CSeq: 1939639 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:[email protected]:5060;transport=TCP> Content-Length: 0 13:37:54.241172 IP (tos 0x0, ttl 64, id 43418, offset 0, flags [none], proto UDP (17), length 1097) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 1069 SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK86e8a99ff6e15f3e72053880758bf877.9856a4e1;received=217.0.17.170;rport=5060 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709 From: <sip:[email protected]>;tag=f18b4044 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd Call-ID: [email protected] CSeq: 1939639 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:[email protected]:5060;transport=TCP> Content-Type: application/sdp Content-Length: 255 v=0 o=root 1580918074 1580918076 IN IP4 79.253.136.104 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 79.253.136.104 t=0 0 m=audio 17212 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 13:37:54.282723 IP (tos 0xc0, ttl 25, id 14239, offset 0, flags [none], proto UDP (17), length 929) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 901 ACK sip:[email protected]:5060;transport=TCP SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK64b752f94c3eb2ddef50d69038a25de8.067eff9c Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bKcabfd322c5154f44ca11d4789d1aa7fdjaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609372294-1570709470 Max-Forwards: 70 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd From: <sip:[email protected]>;tag=f18b4044 Call-ID: [email protected] Contact: <sip:[email protected]:5082;transport=tcp>;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel" CSeq: 1939639 ACK Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE Content-Length: 0 13:37:54.286434 IP (tos 0xc0, ttl 25, id 14256, offset 0, flags [none], proto UDP (17), length 468) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 440 BYE sip:[email protected]:5060;transport=TCP SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK14b26a70d8dbb95a10976126acb08635.224832b1 Max-Forwards: 70 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd From: <sip:[email protected]>;tag=f18b4044 Call-ID: [email protected] Contact: <sip:[email protected]:5060> CSeq: 1939640 BYE Content-Length: 0 13:37:54.286700 IP (tos 0x0, ttl 64, id 43419, offset 0, flags [none], proto UDP (17), length 547) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 519 SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK14b26a70d8dbb95a10976126acb08635.224832b1;received=217.0.17.170;rport=5060 From: <sip:[email protected]>;tag=f18b4044 To: 0900666666 <sip:[email protected]>;tag=as09bca4fd Call-ID: [email protected] CSeq: 1939640 BYE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 13:37:54.339838 IP (tos 0x0, ttl 64, id 43420, offset 0, flags [none], proto UDP (17), length 1064) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 1036 SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b;received=217.0.17.170;rport=5060 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 From: <sip:[email protected];user=phone>;tag=8f233b97 To: <sip:[email protected]:5060>;tag=as77f2fb84 Call-ID: [email protected] CSeq: 1939619 INVITE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Type: application/sdp Content-Length: 253 v=0 o=root 515584563 515584563 IN IP4 79.253.136.104 s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 c=IN IP4 79.253.136.104 t=0 0 m=audio 16240 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=ptime:20 a=sendrecv 13:37:54.384756 IP (tos 0xc0, ttl 25, id 14594, offset 0, flags [none], proto UDP (17), length 890) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 862 ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bKea131109e3bc752ecd42b1bcf6623ebc.e6df8be1 Via: SIP/2.0/TCP 62.156.82.55:5060;branch=z9hG4bK7486879ebc5b30e8bf65ebc351d2e893jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609471157-1129494485 Max-Forwards: 70 To: <sip:[email protected]:5060>;tag=as77f2fb84 From: <sip:[email protected];user=phone>;tag=8f233b97 Call-ID: [email protected] Contact: <sip:[email protected]:5083;transport=tcp>;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel" CSeq: 1939619 ACK Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE Content-Length: 0 13:37:54.385007 IP (tos 0x0, ttl 64, id 43421, offset 0, flags [none], proto UDP (17), length 683) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 655 BYE sip:[email protected]:5083;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 79.253.136.104:5060;branch=z9hG4bK424d4fd6;rport Route: <sip:[email protected]:5060;transport=tcp;lr>,<sip:3zqkv7%1bbaqeoaaaaduansdj97oyoaaaaaytel%[email protected];lr> Max-Forwards: 70 From: <sip:[email protected]:5060>;tag=as77f2fb84 To: <sip:[email protected];user=phone>;tag=8f233b97 Call-ID: [email protected] CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 13:37:54.388625 IP (tos 0xc0, ttl 25, id 14613, offset 0, flags [none], proto UDP (17), length 438) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 410 BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bKf8e5ee59616d6d8a90e1c34092ca7208.dcc6107f Max-Forwards: 70 To: <sip:[email protected]:5060>;tag=as77f2fb84 From: <sip:[email protected];user=phone>;tag=8f233b97 Call-ID: [email protected] Contact: <sip:[email protected]:5060> CSeq: 1939620 BYE Content-Length: 0 13:37:54.388816 IP (tos 0x0, ttl 64, id 43422, offset 0, flags [none], proto UDP (17), length 531) 172.16.0.2.5060 > 217.0.17.170.5060: SIP, length: 503 SIP/2.0 200 OK Via: SIP/2.0/UDP 217.0.17.170:5060;branch=z9hG4bKf8e5ee59616d6d8a90e1c34092ca7208.dcc6107f;received=217.0.17.170;rport=5060 From: <sip:[email protected];user=phone>;tag=8f233b97 To: <sip:[email protected]:5060>;tag=as77f2fb84 Call-ID: [email protected] CSeq: 1939620 BYE Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 13:37:54.404027 IP (tos 0xc0, ttl 25, id 14661, offset 0, flags [none], proto UDP (17), length 391) 217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 363 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 79.253.136.104:5060;rport=5060;branch=z9hG4bK424d4fd6 To: <sip:[email protected];user=phone>;tag=8f233b97 From: <sip:[email protected]:5060>;tag=as77f2fb84 Call-ID: [email protected] Contact: <sip:[email protected]:5060> CSeq: 102 BYE Content-Length: 0 --- 8< snap --- I hope this is still readable... ;-) Best regards Flo > > Peter -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
