According to RFC 3261 page 13 wich is:<br><br> CSeq or Command Sequence contains an integer and a method name. The<br> CSeq number is incremented for each new request within a dialog and<br> is a traditional sequence number.<br>
<br>But when we send reinvite CSeq number is decrementing from 635011974 to 102. Have anyone had this bug ?. <br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 212.150.30.11:5060;branch=z9hG4bKfpfiih206gh16ik5p0o1cb59m7832.1;received=212.150.30.11<br>
From: <<a href="mailto:sip%3A74952201010@212.150.30.11">sip:74952201010@212.150.30.11</a>>;tag=11047689-1283317761800-<br>To: "FUTURE COMM"<<a href="http://sip:78123308802@234.54.143.88:5060">sip:78123308802@234.54.143.88:5060</a>>;tag=as2ad71683<br>
Call-ID: <a href="mailto:BW110921800010910266598393@212.150.30.11">BW110921800010910266598393@212.150.30.11</a><br>CSeq: 635011974 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <<a href="mailto:sip%3A78123308802@234.54.143.88">sip:78123308802@234.54.143.88</a>><br>Content-Type: application/sdp<br>Content-Length: 238<br><br>v=0<br>o=root 1025 1026 IN IP4 234.54.143.88<br>
s=session<br>c=IN IP4 234.54.143.88<br>t=0 0<br>m=audio 10556 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><------------><br>
SIPGW_ROSTEL_FUTURE*CLI><br><--- SIP read from <a href="http://212.150.30.11:5060">212.150.30.11:5060</a> ---><br>ACK <a href="mailto:sip%3A78123308802@234.54.143.88">sip:78123308802@234.54.143.88</a> SIP/2.0<br>
Via: SIP/2.0/UDP 212.150.30.11:5060;branch=z9hG4bKf9mlkq2068v11jk3h4k0.1<br>From: <<a href="mailto:sip%3A74952201010@212.150.30.11">sip:74952201010@212.150.30.11</a>>;tag=11047689-1283317761800-<br>To: "FUTURE COMM"<<a href="http://sip:78123308802@234.54.143.88:5060">sip:78123308802@234.54.143.88:5060</a>>;tag=as2ad71683<br>
Call-ID: <a href="mailto:BW110921800010910266598393@212.150.30.11">BW110921800010910266598393@212.150.30.11</a><br>CSeq: 635011974 ACK<br>Contact: <sip:74952201010@212.150.30.11:5060;transport=udp><br>Max-Forwards: 9<br>
Content-Length: 0<br><br><-------------><br>--- (9 headers 0 lines) ---<br>set_destination: Parsing <sip:74952201010@212.150.30.11:5060;transport=udp> for address/port to send to<br>set_destination: set destination to 212.150.30.11, port 5060<br>
Reliably Transmitting (no NAT) to <a href="http://212.150.30.11:5060">212.150.30.11:5060</a>:<br>INVITE sip:74952201010@212.150.30.11:5060;transport=udp SIP/2.0<br>Via: SIP/2.0/UDP 234.54.143.88:5060;branch=z9hG4bK4b37f023<br>
From: "FUTURE COMM"<<a href="http://sip:78123308802@234.54.143.88:5060">sip:78123308802@234.54.143.88:5060</a>>;tag=as2ad71683<br>To: <<a href="mailto:sip%3A74952201010@212.150.30.11">sip:74952201010@212.150.30.11</a>>;tag=11047689-1283317761800-<br>
Contact: <<a href="mailto:sip%3A78123308802@234.54.143.88">sip:78123308802@234.54.143.88</a>><br>Call-ID: <a href="mailto:BW110921800010910266598393@212.150.30.11">BW110921800010910266598393@212.150.30.11</a><br>CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>X-asterisk-info: SIP re-invite (T38 switchover)<br>Content-Type: application/sdp<br>
Content-Length: 347<br><br>v=0<br>o=root 1025 1027 IN IP4 234.54.143.88<br>s=session<br>c=IN IP4 234.54.143.88<br>t=0 0<br>m=image 4927 udptl t38<br>a=T38FaxVersion:0<br>a=T38MaxBitRate:9600<br>a=T38FaxFillBitRemoval:0<br>
a=T38FaxTranscodingMMR:0<br>a=T38FaxTranscodingJBIG:0<br>a=T38FaxRateManagement:transferredTCF<br>a=T38FaxMaxBuffer:122<br>a=T38FaxMaxDatagram:122<br>a=T38FaxUdpEC:t38UDPRedundancy<br><br>---<br><br>