thr3ads.net - asterisk users - [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes [Mar 2013]

If this information is useful, please help other people find it:
Share via:

Florian Wolters

2013-Mar-21 07:31 UTC

[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

isrlgb at gmail.com

2013-Mar-21 07:50 UTC

head link

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

Try canreinvite=yes in sip trunk

-----Original Message-----
From: Florian Wolters <florian at florian-wolters.de>
Sender: asterisk-users-bounces at lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54 
To: <asterisk-users at lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at 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

Leandro Dardini

2013-Mar-21 08:04 UTC

head link

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013/3/21 Florian Wolters <florian at
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

Zyumbilev, Peter

2013-Mar-21 11:17 UTC

head link

[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
>

Robert Krakora

2013-Mar-21 12:52 UTC

head link

[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 <florian at
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130321/7e6bfdce/attachment.htm>

Florian Wolters

2013-Mar-21 12:56 UTC

head link

[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:0900666666 at 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=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm
	Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843
	Max-Forwards: 70
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	From: <sip:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	Contact:
<sip:p65558t1363868566m240730c3684606s1 at
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:0900666666 at 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=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa
	Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709
	Max-Forwards: 70
	To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
	From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	Contact:
<sip:p65558t1363868566m240730c3684606s3 at
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
	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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	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:0900666666 at 79.253.136.104:5060>
	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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	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:0900666666 at 79.253.136.104:5060>
	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:023468727853 at tel.t-online.de>;tag=f18b4044
	To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	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:0900666666 at 79.253.136.104:5060;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:023468727853 at tel.t-online.de>;tag=f18b4044
	To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	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:0900666666 at 79.253.136.104:5060;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:0900666666 at 79.253.136.104:5060;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:0900666666 at tel.t-online.de>;tag=as09bca4fd
	From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	Contact:
<sip:p65558t1363868566m240730c3684606s3 at
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 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:0900666666 at 79.253.136.104:5060;transport=TCP SIP/2.0
	Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK14b26a70d8dbb95a10976126acb08635.224832b1
	Max-Forwards: 70
	To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
	From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
	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:023468727853 at tel.t-online.de>;tag=f18b4044
	To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
	Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
	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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	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:0900666666 at 79.253.136.104:5060>
	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:0900666666 at 79.253.136.104:5060 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:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	From: <sip:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	Contact:
<sip:p65558t1363868566m240730c3684606s1 at
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 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:p65558t1363868566m240730c3684606s1 at 62.156.80.48:5083;transport=tcp
SIP/2.0
	Via: SIP/2.0/UDP 79.253.136.104:5060;branch=z9hG4bK424d4fd6;rport
	Route:
<sip:DTMTASP01 at
217.0.17.170:5060;transport=tcp;lr>,<sip:3Zqkv7%1bbaqeOaaaaduaNsDJ97OyOaaaaaytel%3a+4923451669387
at hno-esca001--vip-sig.tel.t-online.de;lr>
	Max-Forwards: 70
	From: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	To: <sip:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	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:0900666666 at 79.253.136.104:5060 SIP/2.0
	Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bKf8e5ee59616d6d8a90e1c34092ca7208.dcc6107f
	Max-Forwards: 70
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	From: <sip:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
	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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
	From: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
	Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
	Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
	CSeq: 102 BYE
	Content-Length: 0

--- 8< snap ---

I hope this is still readable... ;-)
Best regards

   Flo


>
> Peter

Jim Lucas

2013-Mar-21 15:19 UTC

head link

[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

Florian Wolters

2013-Mar-22 08:51 UTC

head link

[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

Florian Wolters

2013-Mar-22 09:26 UTC

head link

[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
>
>

Jamie A. Stapleton

2013-Mar-22 17:14 UTC

head link

[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-bounces at lists.digium.com [mailto:asterisk-users-bounces
at 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:0900666666 at 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=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaaaaaiaaaaaaj23clqa3Zqkv7akae3e3wetjnxm
        Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843
        Max-Forwards: 70
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        From: <sip:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        Contact:
<sip:p65558t1363868566m240730c3684606s1 at
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:0900666666 at 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=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaaaaaiaaaaaahr0zo2a3Zqkv7awon0rib4uosfa
        Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709
        Max-Forwards: 70
        To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
        From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        Contact:
<sip:p65558t1363868566m240730c3684606s3 at
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
        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:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        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:0900666666 at 79.253.136.104:5060>
        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:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        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:0900666666 at 79.253.136.104:5060>
        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:023468727853 at tel.t-online.de>;tag=f18b4044
        To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        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:0900666666 at 79.253.136.104:5060;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:023468727853 at tel.t-online.de>;tag=f18b4044
        To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        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:0900666666 at 79.253.136.104:5060;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:0900666666 at 79.253.136.104:5060;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:0900666666 at tel.t-online.de>;tag=as09bca4fd
        From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        Contact:
<sip:p65558t1363868566m240730c3684606s3 at
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 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:0900666666 at 79.253.136.104:5060;transport=TCP SIP/2.0
        Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK14b26a70d8dbb95a10976126acb08635.224832b1
        Max-Forwards: 70
        To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
        From: <sip:023468727853 at tel.t-online.de>;tag=f18b4044
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
        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:023468727853 at tel.t-online.de>;tag=f18b4044
        To: 0900666666 <sip:0900666666 at tel.t-online.de>;tag=as09bca4fd
        Call-ID: 248ef1b5553e5756490d6556573a1fb1 at tel.t-online.de
        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:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        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:0900666666 at 79.253.136.104:5060>
        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:0900666666 at 79.253.136.104:5060 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:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        From: <sip:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        Contact:
<sip:p65558t1363868566m240730c3684606s1 at
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 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:p65558t1363868566m240730c3684606s1 at 62.156.80.48:5083;transport=tcp
SIP/2.0
        Via: SIP/2.0/UDP 79.253.136.104:5060;branch=z9hG4bK424d4fd6;rport
        Route:
<sip:DTMTASP01 at
217.0.17.170:5060;transport=tcp;lr>,<sip:3Zqkv7%1bbaqeOaaaaduaNsDJ97OyOaaaaaytel%3a+4923451669387
at hno-esca001--vip-sig.tel.t-online.de;lr>
        Max-Forwards: 70
        From: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        To: <sip:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        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:0900666666 at 79.253.136.104:5060 SIP/2.0
        Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bKf8e5ee59616d6d8a90e1c34092ca7208.dcc6107f
        Max-Forwards: 70
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        From: <sip:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
        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:+498003301000 at
tel.t-online.de;user=phone>;tag=8f233b97
        To: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        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:+498003301000 at tel.t-online.de;user=phone>;tag=8f233b97
        From: <sip:0900666666 at 79.253.136.104:5060>;tag=as77f2fb84
        Call-ID: 83de2b0c3faf0ef9 at 217.0.17.170
        Contact: <sip:DTMTASP01 at 217.0.17.170:5060>
        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

asterisk users - Mar 2013 - Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

Maybe Matching Threads