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Home » Kerio User Forums » Kerio Operator » Internal & External Call Drop Out 30 Minutes - -HangupCause: Normal Clearing (Internal & External Calls drop at 30 Minutes)
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tomehb

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Hi Kerio,

We currently have a trial version of Kerio Connect, and have also just ordered the full version. However it appears that our calls are dropping after 30 minutes. The Maximum duration of an out going call is set to 6 hours, however all calls drop at 30 minutes.

It would appear that the calls are set to expire after 30 minutes, but this is meant to be refreshed. However for some reason this is not being refreshed?


asterisk[31435]: WARNING[31455]: chan_sip.c:26346 in proc_session_timer: Session-Timer expired - 0010de38-d104-e311-83f1-ae7cc07bf630"


and

from packet dumps, (Session-Expires: 1800;refresher=uac)
./fa/3096/0/

and then from the debug Log when the call is dropped...
[16/Aug/2013 16:14:27] {sip} asterisk[31435]: VERBOSE[5069]: chan_sip.c:4287 in send_request: Reliably Transmitting (no NAT) to 10.99.7.176:1024:#012BYE
 sip:9505<_at_>10.99.7.176:1024 SIP/2.0#015#012Via: SIP/2.0/UDP 10.99.9.2:5060;branch=z9hG4bK6234cb8e#015#012Max-Forwards: 70#015#012
 From: "Tom Buchanan" <sip:4211<_at_>10.99.9.2>;tag=as6b836b8b#015#012
 To: <sip:9505@10.99.7.176:1024>;tag=x051ldkj0x#015#012Call-ID: 309b30a243f8f9c1191639187bf16340<_at_>10.99.9.2:5060#015#012CSeq: 103 BYE#015#012User-Agent: 
 Asterisk PBX#015#012X-Asterisk-HangupCause: Normal Clearing#015#012X-Asterisk-HangupCauseCode: 16#015#012Content-Length: 0#015#012#015#012#012---
 
 [16/Aug/2013 16:14:27] {sip} asterisk[31435]: VERBOSE[5069]: chan_sip.c:4285 in send_request: Reliably Transmitting (NAT) to 10.99.7.3:5060:#012BYE
  sip:3ComCallProcessor<_at_>10.99.7.3 SIP/2.0#015#012Via: SIP/2.0/UDP 10.99.9.2:5060;branch=z9hG4bK19c49ad0;rport#015#012Max-Forwards: 70#015#012
  From: <sip:9505@10.99.9.2>;tag=as362ae68e#015#012To: "Tom Buchanan"<sip:4211<_at_>10.99.7.3;user=phone>;tag=885ee8f8-01d6-0c52-1d82-00e0bb2c4649#015#012Call-ID: 8017c604-f004-e311-aa02-c4dc67aba4f9#015#012CSeq: 102 BYE#015#012
  User-Agent: Asterisk PBX#015#012X-Asterisk-HangupCause: Normal Clearing#015#012X-Asterisk-HangupCauseCode: 16#015#012Content-Length: 0#015#012#015#012#012---
  
  [16/Aug/2013 16:14:27] {sip} asterisk[31435]: VERBOSE[31455]: chan_sip.c:25682 in handle_request_do: #012<--- SIP read from UDP:10.99.7.3:5060 --->#012SIP/2.0 200 OK#012via: 
  SIP/2.0/UDP 10.99.9.2:5060;branch=z9hG4bK19c49ad0;rport#012from: <sip:9505@10.99.9.2>;tag=as362ae68e#012to: "Tom Buchanan"<sip:4211<_at_>10.99.7.3;user=phone>;
  tag=885ee8f8-01d6-0c52-1d82-00e0bb2c4649#012call-id: 8017c604-f004-e311-aa02-c4dc67aba4f9#012cseq: 102 BYE#012date: Fri, 16 Aug 2013 15:14:27 GMT#012contact: 
  <sip:3ComCallProcessor<_at_>10.99.7.3>#012user-agent: 3Com VCX 7210 IP CallProcessor/v7.1.215#012X-Asterisk-HangupCause: Normal Clearing#012X-Asterisk-HangupCauseCode: 
  16#012content-length: 0#012#012<------------->



Does anyone have any ideas?

Thanks

Tom

[Updated on: Sun, 18 August 2013 17:29]

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Vladimir Toncar (Kerio)

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This is a known issue with some SIP carriers who do not handle session timers well. See for example http://forums.kerio.com/t/22862//

As a work-around, you can use a configuration hook file. Contact me directly at vtoncar at kerio dot com for details.

Vladimir
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silars

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The only thing different about this case than mine is that internal calls are dropping at 30 mins as well. I didn't have this problem on internal calls, just external VoIP ITSP calls. VoIP Gateway calls worked fine as well.

Could this be a handset configuration issue?
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Vladimir Toncar (Kerio)

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Hi,

The next step should be to capture the packets at the beginning of the call and at the end. Then use Wireshark/Ethereal to see how the session timers are handled in the SIP signalling.

Vladimir
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