[asterisk-bugs] Asterisk 1.4.21.2 SIP CSeq incrementing problem

[Mohail Timofeev]

According to  RFC 3261  page 13 wich is:

 CSeq or Command Sequence contains an integer and a method name. The
 CSeq number is incremented for each new request within a dialog and
 is a traditional sequence number.

But when we send reinvite CSeq number is decrementing from 635011974 to 102.
Have anyone had this bug ?.

SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.150.30.11:5060
;branch=z9hG4bKfpfiih206gh16ik5p0o1cb59m7832.1;received=212.150.30.11
From: <sip:74952201010 at 212.150.30.11 <sip%3A74952201010 at 212.150.30.11>
>;tag=11047689-1283317761800-
To: "FUTURE COMM"<sip:78123308802 at 234.54.143.88:5060>;tag=as2ad71683
Call-ID: BW110921800010910266598393 at 212.150.30.11
CSeq: 635011974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:78123308802 at 234.54.143.88 <sip%3A78123308802 at 234.54.143.88>>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1025 1026 IN IP4 234.54.143.88
s=session
c=IN IP4 234.54.143.88
t=0 0
m=audio 10556 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
SIPGW_ROSTEL_FUTURE*CLI>
<--- SIP read from 212.150.30.11:5060 --->
ACK sip:78123308802 at 234.54.143.88 <sip%3A78123308802 at 234.54.143.88> SIP/2.0
Via: SIP/2.0/UDP 212.150.30.11:5060;branch=z9hG4bKf9mlkq2068v11jk3h4k0.1
From: <sip:74952201010 at 212.150.30.11 <sip%3A74952201010 at 212.150.30.11>
>;tag=11047689-1283317761800-
To: "FUTURE COMM"<sip:78123308802 at 234.54.143.88:5060>;tag=as2ad71683
Call-ID: BW110921800010910266598393 at 212.150.30.11
CSeq: 635011974 ACK
Contact: <sip:74952201010 at 212.150.30.11:5060;transport=udp>
Max-Forwards: 9
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:74952201010 at 212.150.30.11:5060;transport=udp>
for address/port to send to
set_destination: set destination to 212.150.30.11, port 5060
Reliably Transmitting (no NAT) to 212.150.30.11:5060:
INVITE sip:74952201010 at 212.150.30.11:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 234.54.143.88:5060;branch=z9hG4bK4b37f023
From: "FUTURE COMM"<sip:78123308802 at 234.54.143.88:5060>;tag=as2ad71683
To: <sip:74952201010 at 212.150.30.11 <sip%3A74952201010 at 212.150.30.11>
>;tag=11047689-1283317761800-
Contact: <sip:78123308802 at 234.54.143.88 <sip%3A78123308802 at 234.54.143.88>>
Call-ID: BW110921800010910266598393 at 212.150.30.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 347

v=0
o=root 1025 1027 IN IP4 234.54.143.88
s=session
c=IN IP4 234.54.143.88
t=0 0
m=image 4927 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPRedundancy

---
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-bugs/attachments/20100901/30671a23/attachment.htm