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[asterisk-users] Calls disconnect after 15 minutes | cause=408 ; text="408 Request Timeout"| Asterisk 11.8.1 -


 
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PostPosted: Wed Jul 30, 2014 11:30 am    Post subject: [asterisk-users] Calls disconnect after 15 minutes | cause=4 Reply with quote

We're experiencing an issue where calls disconnect after 15 minutes.  It seems to happen just after Asterisk sends an  update mesage.






RTP is being set up directly.  Asterisk is only in the SIP dialog.


Has anyone experienced this issue? 








4 PRIs inbound, 4 PRIs outbound, asterisk provides switching.






SIP/2.0 200 OK  
 Via: SIP/2.0/UDP 38.XXX.XXX.XXX:5060;branch=z9hG4bK1c4b524f  
 From: <sip:18609700010@38.XXX.XXX.XXX;user=phone>;tag=as23a58665  
 To: "Conference Room" <sip:8009XXXXXX@38.XXX.XXXX.XXX>;tag=1c241709270  
 Call-ID: 2417070873072014102945@38.XXX.XXXX.XXX  
 CSeq: 103 UPDATE  
 Contact: <sip:8009XXXXXX@38.XXX.XXXX.XXX:5060>  
 Supported: em,timer,replaces,path,resource-priority  
 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE  
 Server: Audiocodes-Sip-Gateway-Media Gateway 3200/v.6.40A.063.001  
 Content-Type: application/sdp  Content-Length: 231    v=0  o=AudiocodesGW 241669226 241668902 IN IP4 38.XXX.XXXX.XXX  s=Phone-Call  c=IN IP4 38.XXX.XXXX.XXX  t=0 0  m=audio 6330 RTP/AVP 0 101  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-15  a=ptime:20  a=sendrecv
 
Jul 30 11:00:06 38.XXX.XXXX.XXX 
BYE sip:18609700010@38.XXX.XXX.XXX:5060 SIP/2.0  
Via: SIP/2.0/UDP 38.XXX.XXXX.XXX;branch=z9hG4bKac1497137359  
Max-Forwards: 70  
From: "Conference Room" <sip:8009XXXXXX@38.XXX.XXXX.XXX>;tag=1c241709270  
To: <sip:18609700010@38.XXX.XXX.XXX;user=phone>;tag=as23a58665  
Call-ID: 2417070873072014102945@38.XXX.XXXX.XXX  
CSeq: 2 BYE  
Supported: em,timer,replaces,path,resource-priority  
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE  
User-Agent: Audiocodes-Sip-Gateway-Media Gateway 3200/v.6.40A.063.001  
Reason: SIP ;cause=408 ;text="408 Request Timeout"  Content-Length: 0
Jul 30 11:00:06 38.XXX.XXXX.XXX 
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 38.XXX.XXXX.XXX;branch=z9hG4bKac1497137359;received=38.XXX.XXXX.XXX  
From: "Conference Room" <sip:8009XXXXXX@38.XXX.XXXX.XXX>;tag=1c241709270  
To: <sip:18609700010@38.XXX.XXX.XXX;user=phone>;tag=as23a58665  
Call-ID: 2417070873072014102945@38.XXX.XXXX.XXX  CSeq: 2 BYE  Server: Vantage_SS  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH  Supported: replaces, timer  Content-Length: 0
root@netlog:/logs/38.XXX.XXXX.XXX/2014/07#
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