<--- SIP read from UDP:10.123.101.172:5060 --->
INVITE sip:73625@10.192.15.174:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:9198277929@10.123.101.172:5060;transport=udp;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 67
Supported: 100rel,timer
User-Agent: Huawei SoftX3000 V300R010
Session-Expires: 300
Min-SE: 90
Content-Length: 300
Content-Type: application/sdp
v=0
o=- 70053899 70053899 IN IP4 10.123.101.172
s=SBC call
c=IN IP4 10.123.101.172
t=0 0
m=audio 35686 RTP/AVP 8 0 18 3 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 telephone-event/8000
a=ptime:20
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (16 headers 14 lines) ---
Sending to 10.123.101.172:5060 (NAT)
Sending to 10.123.101.172:5060 (NAT)
Using INVITE request as basis request - isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
Found peer 'T73625' for '9198277929' from 10.123.101.172:5060
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 97
Capabilities: us - (ulaw|gsm|alaw|g722|h263|h264|vp8), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x16f1090 -- Strict RTP learning after remote address set to: 10.123.101.172:35686
Peer audio RTP is at port 10.123.101.172:35686
Peer doesn't provide video
Looking for 73625 in from-trunk (domain 10.192.15.174)
sip_route_dump: route/path hop: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
<--- Transmitting (NAT) to 10.123.101.172:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Length: 0
<------------>
-- Executing [73625@from-trunk:1] Set("SIP/T73625-0000000f", "FROM_DID=73625") in new stack
-- Executing [73625@from-trunk:2] Gosub("SIP/T73625-0000000f", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/T73625-0000000f", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/T73625-0000000f", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/T73625-0000000f", "") in new stack
-- Executing [73625@from-trunk:3] Set("SIP/T73625-0000000f", "CDR(did)=73625") in new stack
-- Executing [73625@from-trunk:4] ExecIf("SIP/T73625-0000000f", "1 ?Set(CALLERID(name)=9198277929)") in new stack
-- Executing [73625@from-trunk:5] Set("SIP/T73625-0000000f", "CHANNEL(musicclass)=default") in new stack
-- Executing [73625@from-trunk:6] Set("SIP/T73625-0000000f", "MOHCLASS=default") in new stack
-- Executing [73625@from-trunk:7] Set("SIP/T73625-0000000f", "CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [73625@from-trunk:8] Set("SIP/T73625-0000000f", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [73625@from-trunk:9] Goto("SIP/T73625-0000000f", "from-did-direct,303,1") in new stack
-- Goto (from-did-direct,303,1)
-- Executing [303@from-did-direct:1] Set("SIP/T73625-0000000f", "RINGTIMER=15") in new stack
-- Executing [303@from-did-direct:2] Macro("SIP/T73625-0000000f", "exten-vm,novm,303,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/T73625-0000000f", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/T73625-0000000f", "TOUCH_MONITOR=1578735477.122") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/T73625-0000000f", "AMPUSER=9198277929") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/T73625-0000000f", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/T73625-0000000f", "1?Set(REALCALLERIDNUM=9198277929)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/T73625-0000000f", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/T73625-0000000f", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/T73625-0000000f", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/T73625-0000000f", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/T73625-0000000f", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/T73625-0000000f", "TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/T73625-0000000f", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/T73625-0000000f", "CALLERID(number)=9198277929") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/T73625-0000000f", "CALLERID(name)=9198277929") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/T73625-0000000f", "CDR(cnum)=9198277929") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/T73625-0000000f", "CDR(cnam)=9198277929") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/T73625-0000000f", "CHANNEL(language)=pr") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/T73625-0000000f", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/T73625-0000000f", "EXTTOCALL=303") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/T73625-0000000f", "PICKUPMARK=303") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/T73625-0000000f", "RT=") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/T73625-0000000f", "sub-record-check,s,1(exten,303,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/T73625-0000000f", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/T73625-0000000f", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/T73625-0000000f", "MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/T73625-0000000f", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/T73625-0000000f", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/T73625-0000000f", "0?Set(REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/T73625-0000000f", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/T73625-0000000f", "REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/T73625-0000000f", "NOW=1578735477") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/T73625-0000000f", "DAY=11") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/T73625-0000000f", "MONTH=01") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/T73625-0000000f", "YEAR=2020") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/T73625-0000000f", "TIMESTR=20200111-130757") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/T73625-0000000f", "FROMEXTEN=9198277929") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/T73625-0000000f", "CALLFILENAME=exten-303-9198277929-20200111-130757-1578735477.122") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/T73625-0000000f", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/T73625-0000000f", "0?callee") in new stack
-- Executing [exten@sub-record-check:2] Set("SIP/T73625-0000000f", "REC_POLICY_MODE=dontcare") in new stack
-- Executing [exten@sub-record-check:3] GotoIf("SIP/T73625-0000000f", "1?caller") in new stack
-- Goto (sub-record-check,exten,10)
-- Executing [exten@sub-record-check:10] Set("SIP/T73625-0000000f", "REC_POLICY_MODE=") in new stack
-- Executing [exten@sub-record-check:11] GosubIf("SIP/T73625-0000000f", "0?record,1(exten,303,9198277929)") in new stack
-- Executing [exten@sub-record-check:12] Return("SIP/T73625-0000000f", "") in new stack
-- Executing [s@macro-exten-vm:7] Macro("SIP/T73625-0000000f", "dial-one,,tr,303") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/T73625-0000000f", "DEXTEN=303") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/T73625-0000000f", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/T73625-0000000f", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/T73625-0000000f", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/T73625-0000000f", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/T73625-0000000f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/T73625-0000000f", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/T73625-0000000f", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/T73625-0000000f", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/T73625-0000000f", "1?next3:continue") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [s@macro-dial-one:24] ExecIf("SIP/T73625-0000000f", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
-- Executing [s@macro-dial-one:25] GotoIf("SIP/T73625-0000000f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/T73625-0000000f", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/T73625-0000000f", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/T73625-0000000f", "DEVICES=303") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/T73625-0000000f", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/T73625-0000000f", "0?Set(DEVICES=03)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/T73625-0000000f", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/T73625-0000000f", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/T73625-0000000f", "THISDIAL=SIP/303") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/T73625-0000000f", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/T73625-0000000f", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/T73625-0000000f", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/T73625-0000000f", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/T73625-0000000f", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/T73625-0000000f", "THISPART2=SIP/303") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/T73625-0000000f", "0?Set(THISPART2=DAHDI/303)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/T73625-0000000f", "NEWDIAL=SIP/303&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/T73625-0000000f", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/T73625-0000000f", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/T73625-0000000f", "THISDIAL=SIP/303") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/T73625-0000000f", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/T73625-0000000f", "DSTRING=SIP/303&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/T73625-0000000f", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/T73625-0000000f", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/T73625-0000000f", "DSTRING=SIP/303") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/T73625-0000000f", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/T73625-0000000f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/T73625-0000000f", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/T73625-0000000f", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/T73625-0000000f", "DB(CALLTRACE/303)=9198277929") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/T73625-0000000f", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/T73625-0000000f", "D_OPTIONS=tr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/T73625-0000000f", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/T73625-0000000f", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/T73625-0000000f", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/T73625-0000000f", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/T73625-0000000f", "CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/T73625-0000000f", "KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/T73625-0000000f", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:38] GotoIf("SIP/T73625-0000000f", "1?godial") in new stack
-- Goto (macro-dial-one,s,43)
-- Executing [s@macro-dial-one:43] Dial("SIP/T73625-0000000f", "SIP/303,,tr") in new stack
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 12094
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.248:5060:
INVITE sip:303@192.168.1.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0719d2a9
Max-Forwards: 70
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>
Contact: <sip:9198277929@192.168.1.55:5060>
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:37:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310
v=0
o=root 1145098186 1145098186 IN IP4 192.168.1.55
s=Asterisk PBX 16.7.0
c=IN IP4 192.168.1.55
t=0 0
m=audio 12094 RTP/AVP 0 3 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
-- Called SIP/303
<--- Transmitting (NAT) to 10.123.101.172:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0719d2a9
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0719d2a9
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>;tag=1444997971
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 102 INVITE
Contact: <sip:303@192.168.1.248:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:303@192.168.1.248:5060>
-- SIP/303-00000010 is ringing
<--- Transmitting (NAT) to 10.123.101.172:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0719d2a9
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>;tag=1444997971
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 102 INVITE
Contact: <sip:303@192.168.1.248:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 260
v=0
o=303 8000 8000 IN IP4 192.168.1.248
s=SIP Call
c=IN IP4 192.168.1.248
t=0 0
m=audio 5020 RTP/AVP 9 8 0 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|gsm|alaw|g722|h263|h264|vp8), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f37cc011820 -- Strict RTP learning after remote address set to: 192.168.1.248:5020
Peer audio RTP is at port 192.168.1.248:5020
sip_route_dump: route/path hop: <sip:303@192.168.1.248:5060>
set_destination: Parsing <sip:303@192.168.1.248:5060> for address/port to send to
set_destination: set destination to 192.168.1.248:5060
Transmitting (no NAT) to 192.168.1.248:5060:
ACK sip:303@192.168.1.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK64e67900
Max-Forwards: 70
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>;tag=1444997971
Contact: <sip:9198277929@192.168.1.55:5060>
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(16.7.0)
Content-Length: 0
-- SIP/303-00000010 answered SIP/T73625-0000000f
Audio is at 12186
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.123.101.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
<------------>
-- Channel SIP/303-00000010 joined 'simple_bridge' basic-bridge <c36bd403-4f7c-486f-8342-74bf299b278e>
-- Channel SIP/T73625-0000000f joined 'simple_bridge' basic-bridge <c36bd403-4f7c-486f-8342-74bf299b278e>
Retransmitting #1 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
> 0x7f37cc011820 -- Strict RTP switching to RTP target address 192.168.1.248:5020 as source
Retransmitting #2 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
Retransmitting #3 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
Retransmitting #4 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
Retransmitting #5 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
> 0x7f37cc011820 -- Strict RTP learning complete - Locking on source address 192.168.1.248:5020
Retransmitting #6 (NAT) to 10.123.101.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.101.172:5060;branch=z9hG4bK7tfsecjldc7j6ljlwiijaclwt;Role=3;Hpt=8f58_16;received=10.123.101.172;rport=5060
Record-Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
From: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
To: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(16.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:73625@178.131.55.84:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 285
v=0
o=root 2038934941 2038934941 IN IP4 178.131.55.84
s=Asterisk PBX 16.7.0
c=IN IP4 178.131.55.84
t=0 0
m=audio 12186 RTP/AVP 0 3 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=maxptime:150
a=sendrecv
[2020-01-11 13:08:15] WARNING[2430]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2020-01-11 13:08:15] WARNING[2430]: chan_sip.c:4143 retrans_pkt: Hanging up call isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Channel SIP/T73625-0000000f left 'simple_bridge' basic-bridge <c36bd403-4f7c-486f-8342-74bf299b278e>
-- Channel SIP/303-00000010 left 'simple_bridge' basic-bridge <c36bd403-4f7c-486f-8342-74bf299b278e>
Scheduling destruction of SIP dialog '47abde487248f5b91e080f456bf7748b@192.168.1.55:5060' in 6400 ms (Method: INVITE)
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/T73625-0000000f' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/T73625-0000000f' in macro 'exten-vm'
set_destination: Parsing <sip:303@192.168.1.248:5060> for address/port to send to
set_destination: set destination to 192.168.1.248:5060
Reliably Transmitting (no NAT) to 192.168.1.248:5060:
BYE sip:303@192.168.1.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK10d17037
Max-Forwards: 70
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>;tag=1444997971
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 103 BYE
User-Agent: IPBX-2.11.0(16.7.0)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
== Spawn extension (from-did-direct, 303, 2) exited non-zero on 'SIP/T73625-0000000f'
-- Executing [h@from-did-direct:1] Macro("SIP/T73625-0000000f", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/T73625-0000000f", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,20)
-- Executing [s@macro-hangupcall:20] NoOp("SIP/T73625-0000000f", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:21] GotoIf("SIP/T73625-0000000f", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,39)
-- Executing [s@macro-hangupcall:39] NoOp("SIP/T73625-0000000f", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:40] GotoIf("SIP/T73625-0000000f", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] NoOp("SIP/T73625-0000000f", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:46] GotoIf("SIP/T73625-0000000f", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,52)
-- Executing [s@macro-hangupcall:52] NoOp("SIP/T73625-0000000f", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:53] GotoIf("SIP/T73625-0000000f", "1?noautomon3") in new stack
-- Goto (macro-hangupcall,s,59)
-- Executing [s@macro-hangupcall:59] NoOp("SIP/T73625-0000000f", "MIXMONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:60] GotoIf("SIP/T73625-0000000f", "1?noautomon4") in new stack
-- Goto (macro-hangupcall,s,62)
-- Executing [s@macro-hangupcall:62] NoOp("SIP/T73625-0000000f", "ONETOUCH_RECFILE=") in new stack
-- Executing [s@macro-hangupcall:63] NoOp("SIP/T73625-0000000f", "CDR recordingfile set to: ") in new stack
-- Executing [s@macro-hangupcall:64] GotoIf("SIP/T73625-0000000f", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,67)
-- Executing [s@macro-hangupcall:67] GotoIf("SIP/T73625-0000000f", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,70)
-- Executing [s@macro-hangupcall:70] GotoIf("SIP/T73625-0000000f", "1?theend") in new stack
-- Goto (macro-hangupcall,s,72)
-- Executing [s@macro-hangupcall:72] AGI("SIP/T73625-0000000f", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK10d17037
From: "9198277929" <sip:9198277929@192.168.1.55>;tag=as207564f8
To: <sip:303@192.168.1.248:5060>;tag=1444997971
Call-ID: 47abde487248f5b91e080f456bf7748b@192.168.1.55:5060
CSeq: 103 BYE
Contact: <sip:303@192.168.1.248:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '47abde487248f5b91e080f456bf7748b@192.168.1.55:5060' Method: INVITE
-- <SIP/T73625-0000000f>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:73] Hangup("SIP/T73625-0000000f", "") in new stack
== Spawn extension (macro-hangupcall, s, 73) exited non-zero on 'SIP/T73625-0000000f' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/T73625-0000000f'
Scheduling destruction of SIP dialog 'isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 10.123.101.172:5060:
BYE sip:9198277929@10.123.101.172:5060;transport=udp;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK21df0f8c;rport
Route: <sip:10.123.101.172:5060;transport=udp;lr;Hpt=8f58_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=14958>
Max-Forwards: 70
From: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
To: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
CSeq: 102 BYE
User-Agent: IPBX-2.11.0(16.7.0)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
<--- SIP read from UDP:10.123.101.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK21df0f8c;received=10.192.15.174;rport=5060
Call-ID: isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000
From: <sip:73625@10.192.15.174;user=phone>;tag=as255f42fc
To: <sip:9198277929@10.192.15.174;user=phone>;tag=1t8mnz2v-CC-35
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'isbc86n4q2ze6q41v4m8136jq802qnj0tej2@SoftX3000' Method: INVITE
Reliably Transmitting (no NAT) to 192.168.1.248:5060:
OPTIONS sip:303@192.168.1.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK342184bc
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as722fd173
To: <sip:303@192.168.1.248:5060>
Contact: <sip:Unknown@192.168.1.55:5060>
Call-ID: 591ebe5153b11fd009bcf38b435206bb@192.168.1.55:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:38:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK342184bc
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as722fd173
To: <sip:303@192.168.1.248:5060>;tag=1374255240
Call-ID: 591ebe5153b11fd009bcf38b435206bb@192.168.1.55:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '591ebe5153b11fd009bcf38b435206bb@192.168.1.55:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.123.101.172:5060:
OPTIONS sip:10.123.101.172 SIP/2.0
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK6a5d1a86;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@178.131.55.84>;tag=as28a8ee50
To: <sip:10.123.101.172>
Contact: <sip:Unknown@178.131.55.84:5060>
Call-ID: 2d7ec15b639c82846f6223794d5e1d20@178.131.55.84:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:38:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:10.123.101.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK6a5d1a86;received=10.192.15.174;rport=5060
Call-ID: 2d7ec15b639c82846f6223794d5e1d20@178.131.55.84:5060
From: "Unknown"<sip:Unknown@178.131.55.84>;tag=as28a8ee50
To: <sip:10.123.101.172>;tag=4emv6868
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2d7ec15b639c82846f6223794d5e1d20@178.131.55.84:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.1:38211:
OPTIONS sip:2000@192.168.10.252:38211 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK11003797;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as65619207
To: <sip:2000@192.168.10.252:38211>
Contact: <sip:Unknown@192.168.1.55:5060>
Call-ID: 2a26e78c249417b22f8e275a2b59aec6@192.168.1.55:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:192.168.1.1:38211 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK11003797;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as65619207
To: <sip:2000@192.168.10.252:38211>;tag=1338682708
Call-ID: 2a26e78c249417b22f8e275a2b59aec6@192.168.1.55:5060
CSeq: 102 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.29
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2a26e78c249417b22f8e275a2b59aec6@192.168.1.55:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.1.1:38211 --->
<------------->
Reliably Transmitting (no NAT) to 192.168.1.248:5060:
OPTIONS sip:303@192.168.1.248:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK27ad39c2
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as37094a9c
To: <sip:303@192.168.1.248:5060>
Contact: <sip:Unknown@192.168.1.55:5060>
Call-ID: 33fce1d71e6644202377af9336a2432e@192.168.1.55:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:39:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:192.168.1.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK27ad39c2
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as37094a9c
To: <sip:303@192.168.1.248:5060>;tag=1703707734
Call-ID: 33fce1d71e6644202377af9336a2432e@192.168.1.55:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.4.138
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '33fce1d71e6644202377af9336a2432e@192.168.1.55:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.123.101.172:5060:
OPTIONS sip:10.123.101.172 SIP/2.0
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK33fc1380;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@178.131.55.84>;tag=as50c4857f
To: <sip:10.123.101.172>
Contact: <sip:Unknown@178.131.55.84:5060>
Call-ID: 5277dd32157735fd217299102a435180@178.131.55.84:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:39:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:10.123.101.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.131.55.84:5060;branch=z9hG4bK33fc1380;received=10.192.15.174;rport=5060
Call-ID: 5277dd32157735fd217299102a435180@178.131.55.84:5060
From: "Unknown"<sip:Unknown@178.131.55.84>;tag=as50c4857f
To: <sip:10.123.101.172>;tag=0jn04j6s
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5277dd32157735fd217299102a435180@178.131.55.84:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.1:38211:
OPTIONS sip:2000@192.168.10.252:38211 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK7cda5e85;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as56c6e7a0
To: <sip:2000@192.168.10.252:38211>
Contact: <sip:Unknown@192.168.1.55:5060>
Call-ID: 28687d7e7593ad782cde62123a3d3646@192.168.1.55:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Sat, 11 Jan 2020 09:39:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:192.168.1.1:38211 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK7cda5e85;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.55>;tag=as56c6e7a0
To: <sip:2000@192.168.10.252:38211>;tag=1530937156
Call-ID: 28687d7e7593ad782cde62123a3d3646@192.168.1.55:5060
CSeq: 102 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.29
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '28687d7e7593ad782cde62123a3d3646@192.168.1.55:5060' Method: OPTIONS
