Positively Optimistic
2014-Jul-30 16:29 UTC
[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
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# -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140730/2cf420c9/attachment.html>
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