[asterisk-users] Calls disconnect after 15 minutes | cause=408 ; text="408 Request Timeout"| Asterisk 11.8.1 --> Audiocodes Mediant 2000 v.6.40A.063.001

[Positively Optimistic]

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 at 38.XXX.XXX.XXX;user=phone>;tag=as23a58665
 To: "Conference Room" <sip:8009XXXXXX at 38.XXX.XXXX.XXX>;tag=1c241709270
 Call-ID: 2417070873072014102945 at 38.XXX.XXXX.XXX
 CSeq: 103 UPDATE
 Contact: <sip:8009XXXXXX at 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 at 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 at 38.XXX.XXXX.XXX>;tag=1c241709270
To: <sip:18609700010 at 38.XXX.XXX.XXX;user=phone>;tag=as23a58665
Call-ID: 2417070873072014102945 at 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 at 38.XXX.XXXX.XXX>;tag=1c241709270
To: <sip:18609700010 at 38.XXX.XXX.XXX;user=phone>;tag=as23a58665
Call-ID: 2417070873072014102945 at 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 at netlog:/logs/38.XXX.XXXX.XXX/2014/07#
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