FIRST 200 OK ... correctly sent to caller

 3(11813) receive_msg: cleaning up
 3(11813) SIP Reply  (status):
 3(11813)  version: <SIP/2.0>
 3(11813)  status:  <200>
 3(11813)  reason:  <OK>
 3(11813) parse_headers: flags=1
 3(11813) Found param type 232, <branch> = <z9hG4bK465c.183e978.1>; state=6
 3(11813) Found param type 234, <received> = <192.168.200.71>; state=16
 3(11813) end of header reached, state=5
 3(11813) parse_headers: Via found, flags=1
 3(11813) parse_headers: this is the first via
 3(11813) header field type 1, name=<Via>, body=<SIP/2.0/UDP 192.168.200.71;branch=z9hG4bK465c.183e978.1;received=192.168.200.71>
 3(11813)  first  via: <SIP/2.0/UDP> <192.168.200.71:(0)> 3(11813) ;<> 3(11813)
 3(11813) exiting parse_msg
 3(11813) After parse_msg...
 3(11813) TIC:tic_response_callback: starting
 3(11813) DEBUG : tic_response_callback : there's no transaction waiting binding
 3(11813) TIC:tic_response_callback:  No operation
 3(11813) forward_reply: found module tm, passing reply to it
 3(11813) DEBUG: t_check: msg id=12 global id=11 T start=0xffffffff
 3(11813) parse_headers: flags=17
 3(11813) Found param type 232, <branch> = <z9hG4bK0a0000960000002e46b3117f00006e8f0000000a>; state=6
 3(11813) Found param type 235, <rport> = <5060>; state=16
 3(11813) end of header reached, state=5
 3(11813) parse_headers: Via found, flags=17
 3(11813) parse_headers: this is the second via
 3(11813) header field type 1, name=<Via>, body=<SIP/2.0/UDP 192.168.200.141;branch=z9hG4bK0a0000960000002e46b3117f00006e8f0000000a;rport=5060>
 3(11813) header field type 512, name=<Record-Route>, body=<<sip:192.168.200.71;ftag=4cb08f2cfe0;lr=on>>
 3(11813) header field type 8, name=<From>, body=<"hp_cesc" <sip:7008@192.168.200.71>;tag=4cb08f2cfe0>
 3(11813) DEBUG: add_param: tag=as6f6964dd
 3(11813) end of header reached, state=29
 3(11813) DEBUG: get_hdr_field: <To> [51]; uri=[sip:grp.commander@192.168.200.71]
 3(11813) DEBUG: to body [<sip:grp.commander@192.168.200.71>]
 3(11813) header field type 4, name=<To>, body=<<sip:grp.commander@192.168.200.71>;tag=as6f6964dd>
 3(11813) header field type 32, name=<Call-ID>, body=<982596E3E4D842A9B30F0F58934E96480x0a000096>
 3(11813) get_hdr_field: cseq <CSeq>: <1> <INVITE>
 3(11813) header field type 16, name=<CSeq>, body=<1 INVITE>
 3(11813) parse_headers: flags=4
 3(11813) DEBUG: t_reply_matching: hash 50532 label 142205825 branch 1
 3(11813) DEBUG: t_reply_matching: reply matched (T=0xb5eada38)!
 3(11813) parse_headers: flags=4
 3(11813) DEBUG: t_check: msg id=12 global id=12 T end=0xb5eada38
 3(11813) DEBUG: reply_received: org. status uas=0, uac[1]=100 local=0 is_invite=1)
 3(11813) DEBUG: reset_timer (group 3, tl=0xb5eadc40)
 3(11813) DEBUG: reset_timer (group 0, tl=0xb5eadc54)
 3(11813) XLOG: xl_print_log: final buffer length 91
 3(11813) ============================================================
 3(11813) [Fri Aug  3 07:29:10 2007]
 3(11813)
 3(11813) XLOG: xl_print_log: final buffer length 27
 3(11813) [192.168.200.71 ## 11813]
 3(11813)
 3(11813) XLOG: xl_print_log: final buffer length 855
 3(11813) SIP/2.0 200 OK
 3(11813) Via: SIP/2.0/UDP 192.168.200.71;branch=z9hG4bK465c.183e978.1;received=192.168.200.71
 3(11813) Via: SIP/2.0/UDP 192.168.200.141;branch=z9hG4bK0a0000960000002e46b3117f00006e8f0000000a;rport=5060
 3(11813) Record-Route: <sip:192.168.200.71;ftag=4cb08f2cfe0;lr=on>
 3(11813) From: "hp_cesc" <sip:7008@192.168.200.71>;tag=4cb08f2cfe0
 3(11813) To: <sip:grp.commander@192.168.200.71>;tag=as6f6964dd
 3(11813) Call-ID: 982596E3E4D842A9B30F0F58934E96480x0a000096
 3(11813) CSeq: 1 INVITE
 3(11813) User-Agent: Asterisk PBX
 3(11813) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 3(11813) Contact: <sip:600@192.168.200.71:6060>
 3(11813) Content-Type: application/sdp
 3(11813) Content-Length: 220
 3(11813) v=0
 3(11813) o=root 11821 11821 IN IP4 192.168.200.71
 3(11813) s=session
 3(11813) c=IN IP4 192.168.200.71
 3(11813) t=0 0
 3(11813) m=audio 24312 RTP/AVP 8 101
 3(11813) a=rtpmap:8 PCMA/8000
 3(11813) a=rtpmap:101 telephone-event/8000
 3(11813) a=fmtp:101 0-16
 3(11813) a=silenceSupp:off - - - -
 3(11813)
 3(11813)
 3(11813) XLOG: xl_print_log: final buffer length 45
 3(11813) RTP Proxy Handle: flagset: globally disabled
 3(11813)
 3(11813) XLOG: xl_print_log: final buffer length 42
 3(11813) LOG: Entering ROUTE RTPPROXY HANDLE BASIC
 3(11813)
 3(11813) XLOG: xl_print_log: final buffer length 48
 3(11813) LOG:  RTPPROXY disabled ... not using RTP Proxy
 3(11813)
 3(11813) ->>>>>>>>> T_code=0, new_code=200
 3(11813) DEBUG: relay_reply: branch=1, save=0, relay=1
 3(11813)  old size: 852, new size: 766
 3(11813) build_res_from_sip_res: copied size: orig:102, new: 16, rest: 750 msg=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.141;branch=z9hG4bK0a0000960000002e46b3117f00006e8f0000000a;rport=5060
Record-Route: <sip:192.168.200.71;ftag=4cb08f2cfe0;lr=on>
From: "hp_cesc" <sip:7008@192.168.200.71>;tag=4cb08f2cfe0
To: <sip:grp.commander@192.168.200.71>;tag=as6f6964dd
Call-ID: 982596E3E4D842A9B30F0F58934E96480x0a000096
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@192.168.200.71:6060>
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 11821 11821 IN IP4 192.168.200.71
s=session
c=IN IP4 192.168.200.71
t=0 0
m=audio 24312 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 3(11813) WARNING:vqm_resize: resize(0) called
 3(11813) DEBUG: update_totag_set: new totag
 3(11813) DEBUG: put on WAIT
 3(11813) DEBUG: add_to_tail_of_timer[2]: 0xb5eada80
 3(11813) DEBUG: reply relayed. buf=0x81e6348: SIP/2.0 2..., shmem=0xb5eb2008: SIP/2.0 2
 3(11813) DEBUG: reset_timer (group 3, tl=0xb5eadb60)
 3(11813) DEBUG: reset_timer (group 0, tl=0xb5eadb74)
 3(11813) DEBUG: reset_timer (group 3, tl=0xb5eadc40)
 3(11813) DEBUG: reset_timer (group 0, tl=0xb5eadc54)
 3(11813) DEBUG: cleanup_uac_timers: RETR/FR timers reset
 3(11813) DEBUG:destroy_avp_list: destroying list (nil)