Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
Jim, Their are many places on the net talking about the 15 minute NAT timeout issue. If you are not using this device, well, maybe it has a similar bug. As I am using a fli4l (Linux Router), this seems to not be the problem. I cannot see any dropped packets or timeouts in the logfiles of this router. Anyway, thanks for the hint. Flo -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
Matthew and list, thanks for your detailed reply. This is a little hard to diagnose without seeing the SIP traffic for the duration of the call. It makes it impossible to tell if the INVITES the provider is sending are related to the call (i.e. have the same Call-ID header), but if they are being sent consistently 15 minutes into every call it may not matter. If the provider is sending you unsolicited INVITES that cause your calls to drop, I'd suggest contacting their customer service and asking them why they are being sent. Does it make sense to have a more detailed tcpdump of the SIP session? If so, how should such a thing been shared without posting too much ASCII text to the list? The provider actually sent you two INVITES in rapid succession with different Call-IDs. Sorry, but I have to give an update about this. After thinking about the dump again, it dawned me. I set up a call forward back to my office phone to test this issue. -.- Should have had a thought about that earlier. Soorrryyy. So I did setup another Extension leading me to a MeetMe conference to at least listen to some MoH while waiting for the 15 Minutes to exceed. This showed the same behaviour. After exactly 15 Minutes, the call is terminated - namely by the provider. The analysis of the dump in Wireshark shows the last 6 SIP packets: 2013-03-21 15:56:50.648141217.0.17.170 = 172.16.0.2Request: INVITE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.648325172.16.0.2 = 217.0.17.170 Status: 100 Trying 2013-03-21 15:56:50.648427172.16.0.2 = 217.0.17.170 Status: 200 OK, with session description 2013-03-21 15:56:50.731436217.0.17.170 = 172.16.0.2Request: ACK sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735426217.0.17.170 = 172.16.0.2Request: BYE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735590172.16.0.2 = 217.0.17.170 Status: 200 OK (manually copied that from the Wireshark window). This looks to me as if the provider for some reason does an INVITE after 15 Minutes, that is not correctly handled by my Asterisk. Is there any timer inside the SIP protocol, that may be aged by 15 Minutes? Or should I have a deeper look at the SIP packets? Best regards Flo -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
Hi List, Try canreinvite=yes in sip trunk This did not make any difference... -.- -Original Message- Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
On Mar 22, 2013, at 5:22 AM, Florian Wolters flor...@florian-wolters.de wrote: So I did setup another Extension leading me to a MeetMe conference to at least listen to some MoH while waiting for the 15 Minutes to exceed. This showed the same behaviour. After exactly 15 Minutes, the call is terminated - namely by the provider. The analysis of the dump in Wireshark shows the last 6 SIP packets: 2013-03-21 15:56:50.648141217.0.17.170 = 172.16.0.2Request: INVITE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.648325172.16.0.2 = 217.0.17.170 Status: 100 Trying 2013-03-21 15:56:50.648427172.16.0.2 = 217.0.17.170 Status: 200 OK, with session description 2013-03-21 15:56:50.731436217.0.17.170 = 172.16.0.2Request: ACK sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735426217.0.17.170 = 172.16.0.2Request: BYE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735590172.16.0.2 = 217.0.17.170 Status: 200 OK (manually copied that from the Wireshark window). This looks to me as if the provider for some reason does an INVITE after 15 Minutes, that is not correctly handled by my Asterisk. Is there any timer inside the SIP protocol, that may be aged by 15 Minutes? Or should I have a deeper look at the SIP packets? Sip session timers? http://doxygen.asterisk.org/trunk/sip_session_timers.html -d -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
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: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] 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:090066@79.253.136.104:5060 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=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaiaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 Max-Forwards: 70 To: sip:090066@79.253.136.104:5060;tag=as77f2fb84 From: sip:+498003301...@tel.t-online.de;user=phone;tag=8f233b97 Call-ID: 83de2b0c3faf0ef9@217.0.17.170 Contact: sip:p65558t1363868566m240730c3684606s1@62.156.80.48:5083;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:090066@79.253.136.104:5060;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=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaiaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709 Max-Forwards: 70 To: 090066 sip:0900666...@tel.t-online.de;tag=as09bca4fd From: sip:023468727...@tel.t-online.de;tag=f18b4044 Call-ID: 248ef1b5553e5756490d6556573a1...@tel.t-online.de Contact: sip:p65558t1363868566m240730c3684606s3@62.156.80.48:5082;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
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
Florian Wolters wrote: Does it make sense to have a more detailed tcpdump of the SIP session? If so, how should such a thing been shared without posting too much ASCII text to the list? SIP sessions are generally small enough to post right to the list. Otherwise, you can put them up on a site like pastebin.com and provide the link. So I did setup another Extension leading me to a MeetMe conference to at least listen to some MoH while waiting for the 15 Minutes to exceed. This showed the same behaviour. After exactly 15 Minutes, the call is terminated - namely by the provider. The analysis of the dump in Wireshark shows the last 6 SIP packets: 2013-03-21 15:56:50.648141217.0.17.170 = 172.16.0.2Request: INVITE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.648325172.16.0.2 = 217.0.17.170 Status: 100 Trying 2013-03-21 15:56:50.648427172.16.0.2 = 217.0.17.170 Status: 200 OK, with session description 2013-03-21 15:56:50.731436217.0.17.170 = 172.16.0.2Request: ACK sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735426217.0.17.170 = 172.16.0.2Request: BYE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735590172.16.0.2 = 217.0.17.170 Status: 200 OK (manually copied that from the Wireshark window). This looks to me as if the provider for some reason does an INVITE after 15 Minutes, that is not correctly handled by my Asterisk. Is there any timer inside the SIP protocol, that may be aged by 15 Minutes? Or should I have a deeper look at the SIP packets? This is where a full SIP trace that includes the messages used to setup the call in the first place would be helpful. I haven't seen anything related to session timers in what you've posted so far, but they may have been negotiated when the call was established. Regardless, your calls are consistently dropping at 15 minutes and you've shown that it's caused by the provider sending an INVITE, waiting for the OK, and then sending a BYE. You have enough to go to them and ask why it's happening. Even if it's something in your Asterisk configuration, they are initiating the hangup and should be able to tell you why. If they can't or won't help you troubleshoot this problem then I'd seriously consider looking for a new provider. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
Try canreinvite=yes in sip trunk -Original Message- From: Florian Wolters flor...@florian-wolters.de Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 21 Mar 2013 08:31:54 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- 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
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
2013/3/21 Florian Wolters flor...@florian-wolters.de: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo I think it is important to know the reason the call is disconnected. Start checking who is sending the BYE and if before the BYE there is other weird packets, like retry of packet sending ... A simple tcpdump can help explain all the mistery. Leandro -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
I had this exact problem with my voip provider a few years ago. It was disconnecting at exactly 5 minutes. 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. Peter On 21/03/2013 09:31, Florian Wolters wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- 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
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
I am having the same problem with Asterisk 11.2.0 and Linphone and it is exactly 15 minutes and occurring with SIP running on our LAN. On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- 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 -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax -- _ -- 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] 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:090066@79.253.136.104:5060 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=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaiaaj23clqa3Zqkv7akae3e3wetjnxm Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843 Max-Forwards: 70 To: sip:090066@79.253.136.104:5060;tag=as77f2fb84 From: sip:+498003301...@tel.t-online.de;user=phone;tag=8f233b97 Call-ID: 83de2b0c3faf0ef9@217.0.17.170 Contact: sip:p65558t1363868566m240730c3684606s1@62.156.80.48:5083;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:090066@79.253.136.104:5060;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=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaiaahr0zo2a3Zqkv7awon0rib4uosfa Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709 Max-Forwards: 70 To: 090066 sip:0900666...@tel.t-online.de;tag=as09bca4fd From: sip:023468727...@tel.t-online.de;tag=f18b4044 Call-ID: 248ef1b5553e5756490d6556573a1...@tel.t-online.de Contact: sip:p65558t1363868566m240730c3684606s3@62.156.80.48:5082;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
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
Florian Wolters wrote: 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. Florian, This is a little hard to diagnose without seeing the SIP traffic for the duration of the call. It makes it impossible to tell if the INVITES the provider is sending are related to the call (i.e. have the same Call-ID header), but if they are being sent consistently 15 minutes into every call it may not matter. If the provider is sending you unsolicited INVITES that cause your calls to drop, I'd suggest contacting their customer service and asking them why they are being sent. The provider actually sent you two INVITES in rapid succession with different Call-IDs. To keep this simple, I'll use the following shorthand: Call 1 = Call-ID 83de2b0c3faf0ef9@217.0.17.170 Call 2 = Call-ID 248ef1b5553e5756490d6556573a1...@tel.t-online.de Call 1 is terminated with a BYE from Asterisk immediately after it gets the ACK from the provider. The provider tried to terminate it immediately with its own BYE, but it lost the race. This results in the Call/Transaction Does Not Exist message at the end of that dialog. Call 2 is terminated with a BYE from the provider immediately after they ACK the OK from Asterisk. As I said above, I'd start out by asking the provider why they are sending these INVITES in the first place. Here is the simple timeline derived from your SIP trace that I worked from: Call 1 Call 2 13:37:54.240304 INVITE From Provider to Asterisk 13:37:54.240497 INVITE From Provider to Asterisk 13:37:54.240593 Trying From Asterisk to Provider 13:37:54.240752 OK From Asterisk to Provider 13:37:54.240976 Trying From Asterisk to Provider 13:37:54.241172 OK From Asterisk to Provider 13:37:54.282723 ACK From Provider to Asterisk 13:37:54.286434 BYE From Provider to Asterisk 13:37:54.286700 OK From Asterisk to Provider 13:37:54.339838 OK From Asterisk to Provider 13:37:54.384756 ACK From Provider to Asterisk 13:37:54.385007 BYE From Asterisk to Provider 13:37:54.388625 BYE From Provider to Asterisk 13:37:54.388816 OK From Asterisk to Provider 13:37:54.404027 Call/Transaction Does Not Exist From Provider to Asterisk Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk disconnecting SIP Calls after 15 Minutes
On 3/21/2013 12:31 AM, Florian Wolters wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo Florian, As both an VoIP provider and phone system vendor, I had this same problem 2 years ago. In my situation, it turned out that it was nothing to do with either the Asterisk box or the provider. The problem was with a router that we had terminating our T1 connection. As an ISP we provide T1's to many customers and we provide the router as well. In this specific case, the customer purchased a data T1 connection with QoS (sip and rtp) then purchased our IP asterisk phone system with SIP trunks from us as well. The way we found this issue was by switching our the T1 router. Turns out that it fixed the problem. Exact same configuration was on each router. So we started scratching our heads... We then looked at the firmware of the two routers and found that they were different. We provide Cisco 26XX routers. Their are many places on the net talking about the 15 minute NAT timeout issue. If you are not using this device, well, maybe it has a similar bug. -- Jim Lucas -- _ -- 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
