rendered paste body<--- SIP read from UDP:192.168.3.5:49466 --->
<------------->
Reliably Transmitting (NAT) to 192.168.3.5:49466:
OPTIONS sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK344282cc;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.3.11>;tag=as1e67f3af
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
Contact: <sip:Unknown@192.168.3.11:5060>
Call-ID: 3c2313bd5e205b9d4ef04a026369e83c@192.168.3.11:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Mon, 02 May 2011 23:31:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK344282cc;rport=5060
Contact: <sip:192.168.3.5:49466>
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=f0f0fb1e
From: "Unknown"<sip:Unknown@192.168.3.11>;tag=as1e67f3af
Call-ID: 3c2313bd5e205b9d4ef04a026369e83c@192.168.3.11:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces, eventlist
User-Agent: Bria 3 release 3.1 stamp 58312
Allow-Events: hold, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '3c2313bd5e205b9d4ef04a026369e83c@192.168.3.11:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.90:5060:
OPTIONS sip:192.168.1.90 SIP/2.0
Via: SIP/2.0/UDP 201.130.102.5:5060;branch=z9hG4bK60ea11fc
Max-Forwards: 70
From: "Unknown" <sip:Unknown@201.130.102.5>;tag=as148c94cd
To: <sip:192.168.1.90>
Contact: <sip:Unknown@201.130.102.5:5060>
Call-ID: 5e9169cc119a1e0d72274ec3469ad0f9@201.130.102.5:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Mon, 02 May 2011 23:31:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.90:5060 --->
SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP 201.130.102.5:5060;branch=z9hG4bK60ea11fc;received=172.23.10.37;rport=5060
To: <sip:192.168.1.90>
From: "Unknown"<sip:Unknown@201.130.102.5>;tag=as148c94cd
Call-ID: 5e9169cc119a1e0d72274ec3469ad0f9@201.130.102.5:5060
CSeq: 102 OPTIONS
User-Agent: ENSR3.0.65.6-IS2
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5e9169cc119a1e0d72274ec3469ad0f9@201.130.102.5:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.1.90:5060 --->
INVITE sip:6642325002@172.23.10.37:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1
Max-Forwards: 34
Contact: <sip:6653757125@192.168.1.90:5060>
To: <sip:6642325002@172.23.10.37;noa=national>
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Content-Type: application/sdp
Date: Mon, 02 May 2011 23:32:01 GMT
User-Agent: ENSR3.0.65.6-IS2-RMRG600-RG701-CPO4377
Content-Length: 295
v=0
o=- 1205969458 1205969458 IN IP4 192.168.1.90
s=ENSResip
c=IN IP4 192.168.1.95
t=0 0
m=audio 13348 RTP/AVP 8 18 0 4 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Sending to 192.168.1.90:5060 (no NAT)
Using INVITE request as basis request - 4dbf3ef10000111500000026b932b0f7@ens.com
Found peer 'GtelSIP' for '6653757125' from 192.168.1.90:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.95:13348
Looking for 6642325002 in from-trunk (domain 172.23.10.37:5060)
list_route: hop: <sip:6653757125@192.168.1.90:5060>
<--- Transmitting (no NAT) to 192.168.1.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Length: 0
<------------>
-- Executing [6642325002@from-trunk:1] Set("SIP/GtelSIP-00000029", "__FROM_DID=6642325002") in new stack
-- Executing [6642325002@from-trunk:2] Gosub("SIP/GtelSIP-00000029", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/GtelSIP-00000029", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/GtelSIP-00000029", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/GtelSIP-00000029", "") in new stack
-- Executing [6642325002@from-trunk:3] ExecIf("SIP/GtelSIP-00000029", "1 ?Set(CALLERID(name)=6653757125)") in new stack
-- Executing [6642325002@from-trunk:4] Set("SIP/GtelSIP-00000029", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [6642325002@from-trunk:5] Set("SIP/GtelSIP-00000029", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [6642325002@from-trunk:6] Goto("SIP/GtelSIP-00000029", "from-did-direct,7687,1") in new stack
-- Goto (from-did-direct,7687,1)
-- Executing [7687@from-did-direct:1] Macro("SIP/GtelSIP-00000029", "exten-vm,7687,7687") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/GtelSIP-00000029", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/GtelSIP-00000029", "AMPUSER=6653757125") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/GtelSIP-00000029", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/GtelSIP-00000029", "1?Set(REALCALLERIDNUM=6653757125)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/GtelSIP-00000029", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/GtelSIP-00000029", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/GtelSIP-00000029", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/GtelSIP-00000029", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/GtelSIP-00000029", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/GtelSIP-00000029", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/GtelSIP-00000029", "CALLERID(number)=6653757125") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/GtelSIP-00000029", "CALLERID(name)=6653757125") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/GtelSIP-00000029", "Using CallerID "6653757125" <6653757125>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/GtelSIP-00000029", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/GtelSIP-00000029", "VMBOX=7687") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/GtelSIP-00000029", "__EXTTOCALL=7687") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/GtelSIP-00000029", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/GtelSIP-00000029", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/GtelSIP-00000029", "RT=20") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/GtelSIP-00000029", "record-enable,7687,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/GtelSIP-00000029", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/GtelSIP-00000029", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/GtelSIP-00000029", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/GtelSIP-00000029", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/GtelSIP-00000029", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/GtelSIP-00000029", "dial-one,20,tr,7687") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/GtelSIP-00000029", "DEXTEN=7687") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/GtelSIP-00000029", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/GtelSIP-00000029", "0?screen,1") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/GtelSIP-00000029", "0?cf,1") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/GtelSIP-00000029", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/GtelSIP-00000029", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/GtelSIP-00000029", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/GtelSIP-00000029", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/GtelSIP-00000029", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/GtelSIP-00000029", "1?next3:continue") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [s@macro-dial-one:24] ExecIf("SIP/GtelSIP-00000029", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
-- Executing [s@macro-dial-one:25] GotoIf("SIP/GtelSIP-00000029", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/GtelSIP-00000029", "1?dstring,1:dlocal,1") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/GtelSIP-00000029", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/GtelSIP-00000029", "DEVICES=7687") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/GtelSIP-00000029", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/GtelSIP-00000029", "0?Set(DEVICES=687)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/GtelSIP-00000029", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/GtelSIP-00000029", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/GtelSIP-00000029", "THISDIAL=SIP/7687") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/GtelSIP-00000029", "1?zap2dahdi,1") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/GtelSIP-00000029", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/GtelSIP-00000029", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/GtelSIP-00000029", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/GtelSIP-00000029", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/GtelSIP-00000029", "THISPART2=SIP/7687") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/GtelSIP-00000029", "0?Set(THISPART2=DAHDI/7687)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/GtelSIP-00000029", "NEWDIAL=SIP/7687&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/GtelSIP-00000029", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/GtelSIP-00000029", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/GtelSIP-00000029", "THISDIAL=SIP/7687") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/GtelSIP-00000029", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/GtelSIP-00000029", "DSTRING=SIP/7687&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/GtelSIP-00000029", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/GtelSIP-00000029", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/GtelSIP-00000029", "DSTRING=SIP/7687") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/GtelSIP-00000029", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/GtelSIP-00000029", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/GtelSIP-00000029", "1?skiptrace") in new stack
-- Goto (macro-dial-one,s,30)
-- Executing [s@macro-dial-one:30] Set("SIP/GtelSIP-00000029", "D_OPTIONS=tr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/GtelSIP-00000029", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/GtelSIP-00000029", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/GtelSIP-00000029", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/GtelSIP-00000029", "0?qwait,1") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/GtelSIP-00000029", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/GtelSIP-00000029", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] Dial("SIP/GtelSIP-00000029", "SIP/7687,20,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 5060
Video is at 192.168.3.11:5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1000 (g722) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.3.5:49466:
INVITE sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport
Max-Forwards: 70
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
Contact: <sip:6653757125@192.168.3.11:5060>
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Mon, 02 May 2011 23:32:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 2074262084 2074262084 IN IP4 192.168.3.11
s=Asterisk PBX 1.8.3.2
c=IN IP4 192.168.3.11
b=CT:384
t=0 0
m=audio 16026 RTP/AVP 8 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17408 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
---
-- Called 7687
<--- Transmitting (no NAT) to 192.168.1.90:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Length: 0
<------------>
Retransmitting #1 (NAT) to 192.168.3.5:49466:
INVITE sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport
Max-Forwards: 70
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
o: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
Contact: <sip:6653757125@192.168.3.11:5060>
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Mon, 02 May 2011 23:32:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 2074262084 2074262084 IN IP4 192.168.3.11
s=Asterisk PBX 1.8.3.2
c=IN IP4 192.168.3.11
b=CT:384
t=0 0
m=audio 16026 RTP/AVP 8 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17408 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
---
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport=5060
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport=5060
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport=5060
Contact: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
From: "6653757125"<sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
User-Agent: Bria 3 release 3.1 stamp 58312
Allow-Events: hold, talk
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/7687-0000002a is ringing
<--- Transmitting (no NAT) to 192.168.1.90:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK550e58be;rport=5060
Contact: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
From: "6653757125"<sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: Bria 3 release 3.1 stamp 58312
Content-Length: 570
v=0
o=- 1304379128008834 1 IN IP4 192.168.3.5
s=Counterpath Bria 3.1
c=IN IP4 192.168.3.5
t=0 0
a=ice-ufrag:6ae260
a=ice-pwd:f7878096675fe4f6b81cc65ed427bd1c
m=audio 50054 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.3.5 50054 typ host
a=candidate:1 2 UDP 659134 192.168.3.5 50055 typ host
m=video 58206 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=1
a=inactive
a=candidate:1 1 UDP 659136 192.168.3.5 58206 typ host
a=candidate:1 2 UDP 659134 192.168.3.5 58207 typ host
<------------->
--- (12 headers 19 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x8100a (gsm|alaw|g722|h263), peer - audio=0x8 (alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80008 (alaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.5:50054
Peer video RTP is at port 192.168.3.5:58206
list_route: hop: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
set_destination: Parsing <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9> for address/port to send to
set_destination: set destination to 192.168.3.5:49466
Transmitting (NAT) to 192.168.3.5:49466:
ACK sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK077356fd;rport
Max-Forwards: 70
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
Contact: <sip:6653757125@192.168.3.11:5060>
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.3.2)
Content-Length: 0
---
-- SIP/7687-0000002a answered SIP/GtelSIP-00000029
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.3.5:49466 --->
<------------->
Retransmitting #2 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #6 (no NAT) to 192.168.1.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK2c0a80160ll6e6497INV47e1a2324dbf3ef1;received=192.168.1.90
From: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
To: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 657701 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:6642325002@201.130.102.5:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2071425833 2071425833 IN IP4 201.130.102.5
s=Asterisk PBX 1.8.3.2
c=IN IP4 201.130.102.5
t=0 0
m=audio 12546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Executing [h@macro-dial-one:1] Macro("SIP/GtelSIP-00000029", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/GtelSIP-00000029", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/GtelSIP-00000029", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/GtelSIP-00000029", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/GtelSIP-00000029", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/GtelSIP-00000029' in macro 'hangupcall'
Scheduling destruction of SIP dialog '677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9> for address/port to send to
set_destination: set destination to 192.168.3.5:49466
Reliably Transmitting (NAT) to 192.168.3.5:49466:
BYE sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK48b2d75a;rport
Max-Forwards: 70
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.3.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/GtelSIP-00000029' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/GtelSIP-00000029' in macro 'exten-vm'
== Spawn extension (from-did-direct, 7687, 1) exited non-zero on 'SIP/GtelSIP-00000029'
Scheduling destruction of SIP dialog '4dbf3ef10000111500000026b932b0f7@ens.com' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:6653757125@192.168.1.90:5060> for address/port to send to
set_destination: set destination to 192.168.1.90:5060
Reliably Transmitting (no NAT) to 192.168.1.90:5060:
BYE sip:6653757125@192.168.1.90:5060 SIP/2.0
Via: SIP/2.0/UDP 201.130.102.5:5060;branch=z9hG4bK36832064
Max-Forwards: 70
From: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
To: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.3.2)
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.130.102.5:5060;branch=z9hG4bK36832064;received=172.23.10.37;rport=5060
To: <sip:6653757125@192.168.1.90;noa=national>;tag=47e1a232-co6577-INS002
From: <sip:6642325002@172.23.10.37;noa=national>;tag=as706a9b7e
Call-ID: 4dbf3ef10000111500000026b932b0f7@ens.com
CSeq: 102 BYE
User-Agent: ENSR3.0.65.6-IS2-RMRG600-RG701-CPO4377
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4dbf3ef10000111500000026b932b0f7@ens.com' Method: INVITE
Retransmitting #1 (NAT) to 192.168.3.5:49466:
BYE sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK48b2d75a;rport
Max-Forwards: 70
From: "6653757125" <sip:6653757125@192.168.3.11>;tag=as4404a265
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.3.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK48b2d75a;rport=5060
Contact: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
From: "6653757125"<sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 103 BYE
User-Agent: Bria 3 release 3.1 stamp 58312
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060' Method: INVITE
<--- SIP read from UDP:192.168.3.5:49466 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK48b2d75a;rport=5060
Contact: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>
To: <sip:7687@192.168.3.5:49466;rinstance=9059ca76ee50c6c9>;tag=245f9f04
From: "6653757125"<sip:6653757125@192.168.3.11>;tag=as4404a265
Call-ID: 677885a02a1ceaae16a5a8676b3ee4ec@192.168.3.11:5060
CSeq: 103 BYE
User-Agent: Bria 3 release 3.1 stamp 58312
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
pbx*CLI> sip debug off
No such command 'sip debug off' (type 'core show help sip debug off' for other possible commands)
pbx*CLI> exit
root@pbx:/etc/sysconfig/network-scripts $