Hello!
Excuse me, i know there are a lot of documentation about NATing SIP traffic, i read a lot and spend some days for debugging.
I have asterisk server behind NAT and extension is behind NAT too. I forwarded all ports and I tried to use “NC -u” to be sure that port is forwarded.
The problem is that asterisk tries to send packets to local address of extension that is behind NAT through internet.
I found some articles like this. It describes my problem, but doesn’t resolves it.
Pbx server address: 192.168.127.6/30 (i think this may be the problem). It’s a VirtualBOX virtual container.
Main server extip: ex.ter.nal.ip
extension local IP: 192.168.1.108
configurations:
sip.conf
[general]
…
nat=yes
externip=ex.ter.nal.ip
localnet=192.168.127.4/255.255.255.252
…
sip_general_additional.conf
vmexten=*97
accept_outofcall_messages=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.7.0)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=speex
callevents=no
jbenable=no
minexpiry=60
maxcallbitrate=384
maxexpiry=3600
notifyhold=yes
notifyringing=yes
registerattempts=0
registertimeout=20
rtpholdtimeout=300
rtpkeepalive=0
rtptimeout=30
srvlookup=yes
allowguest=yes
checkmwi=10
defaultexpiry=120
videosupport=no
canreinvite=no
g726nonstandard=no
nat=yes
externip=ex.ter.nal.ip
localnet=192.168.127.4/255.255.255.252
debug
*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:192.168.127.5:27537 —>
INVITE sip:[email protected] SIP/2.0
Call-ID: [email protected]
CSeq: 6869 INVITE
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK3472d50abf5e38215d45e6f207d9ad63323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Type: application/sdp
Content-Length: 299
v=0
o=- 1390340138487 1390340138489 IN IP4 192.168.1.108
s=-
c=IN IP4 192.168.1.108
t=0 0
m=audio 28832 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
— (10 headers 13 lines) —
Sending to 192.168.127.5:27537 (NAT)
Sending to 192.168.127.5:27537 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1001’ for ‘1001’ from 192.168.127.5:27537
<— Reliably Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK3472d50abf5e38215d45e6f207d9ad63323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as5fc82be7
Call-ID: [email protected]
CSeq: 6869 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1ac21722"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:192.168.127.5:27537 —>
ACK sip:[email protected] SIP/2.0
Call-ID: [email protected]
Max-Forwards: 70
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as5fc82be7
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK3472d50abf5e38215d45e6f207d9ad63323437;rport
CSeq: 6869 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:192.168.127.5:27537 —>
INVITE sip:[email protected]:5060 SIP/2.0
Call-ID: [email protected]
CSeq: 6870 INVITE
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK7c5a5122d151c781f9c8e85e6cffdcba323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Type: application/sdp
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1ac21722”,uri=“sip:[email protected]:5060”,response=“ec02e28b7915851587336157dd32b427”,algorithm=MD5
Content-Length: 299
v=0
o=- 1390340138487 1390340138489 IN IP4 192.168.1.108
s=-
c=IN IP4 192.168.1.108
t=0 0
m=audio 28832 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
— (11 headers 13 lines) —
Sending to 192.168.127.5:27537 (NAT)
Sending to 192.168.127.5:27537 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1001’ for ‘1001’ from 192.168.127.5:27537
<— Reliably Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK7c5a5122d151c781f9c8e85e6cffdcba323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as3086b828
Call-ID: [email protected]
CSeq: 6870 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="122cc87d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:192.168.127.5:27537 —>
ACK sip:[email protected]:5060 SIP/2.0
Call-ID: [email protected]
Max-Forwards: 70
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as3086b828
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK7c5a5122d151c781f9c8e85e6cffdcba323437;rport
CSeq: 6870 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:192.168.127.5:27537 —>
INVITE sip:[email protected]:5060 SIP/2.0
Call-ID: [email protected]
CSeq: 6871 INVITE
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Type: application/sdp
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“122cc87d”,uri=“sip:[email protected]:5060”,response=“fd7d91594ca1498378d3eeab45f04f58”,algorithm=MD5
Content-Length: 299
v=0
o=- 1390340138487 1390340138489 IN IP4 192.168.1.108
s=-
c=IN IP4 192.168.1.108
t=0 0
m=audio 28832 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
— (11 headers 13 lines) —
Sending to 192.168.127.5:27537 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1001’ for ‘1001’ from 192.168.127.5:27537
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 127
Found unknown media description format GSM-EFR for ID 96
Found unknown media description format AMR for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 127
Capabilities: us - (gsm|ulaw|alaw|speex), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.108:28832
Looking for 89261115511 in from-internal (domain ex.ter.nal.ip)
list_route: hop: sip:[email protected]:33574;transport=udp
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
– Executing [89261115511@from-internal:1] Macro(“SIP/1001-00000019”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/1001-00000019”, “TOUCH_MONITOR=1390340141.25”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/1001-00000019”, “AMPUSER=1001”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/1001-00000019”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/1001-00000019”, “1?Set(REALCALLERIDNUM=1001)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/1001-00000019”, “AMPUSER=1001”) in new stack
– Executing [s@macro-user-callerid:6] Set(“SIP/1001-00000019”, “AMPUSERCIDNAME=Alex”) in new stack
– Executing [s@macro-user-callerid:7] GotoIf(“SIP/1001-00000019”, “0?report”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/1001-00000019”, “AMPUSERCID=1001”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/1001-00000019”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/1001-00000019”, “CALLERID(all)=“Alex” <1001>”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/1001-00000019”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“SIP/1001-00000019”, “1?Set(GROUP(concurrency_limit)=1001)”) in new stack
– Executing [s@macro-user-callerid:13] GosubIf(“SIP/1001-00000019”, “7?sub-ccss,s,1(from-internal,89261115511)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/1001-00000019”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/1001-00000019”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/1001-00000019”, “0?monitor_config,1(from-internal,89261115511):monitor_default,1(from-internal,89261115511)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/1001-00000019”, “0?is_exten”) in new stack
– Executing [monitor_default@sub-ccss:2] StackPop(“SIP/1001-00000019”, “”) in new stack
– Executing [monitor_default@sub-ccss:3] Return(“SIP/1001-00000019”, “FALSE”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“SIP/1001-00000019”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,27)
– Executing [s@macro-user-callerid:27] Set(“SIP/1001-00000019”, “CALLERID(number)=1001”) in new stack
– Executing [s@macro-user-callerid:28] Set(“SIP/1001-00000019”, “CALLERID(name)=Alex”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/1001-00000019”, “CDR(cnum)=1001”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/1001-00000019”, “CDR(cnam)=Alex”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/1001-00000019”, “CHANNEL(language)=en”) in new stack
– Executing [89261115511@from-internal:2] Set(“SIP/1001-00000019”, “MOHCLASS=default”) in new stack
– Executing [89261115511@from-internal:3] ExecIf(“SIP/1001-00000019”, “1?Set(TRUNKCIDOVERRIDE=1504)”) in new stack
– Executing [89261115511@from-internal:4] Set(“SIP/1001-00000019”, “_NODEST=”) in new stack
– Executing [89261115511@from-internal:5] Gosub(“SIP/1001-00000019”, “sub-record-check,s,1(out,89261115511,)”) in new stack
– Executing [s@sub-record-check:1] Set(“SIP/1001-00000019”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:2] GotoIf(“SIP/1001-00000019”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [s@sub-record-check:7] Set(“SIP/1001-00000019”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:8] GotoIf(“SIP/1001-00000019”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [s@sub-record-check:11] ExecIf(“SIP/1001-00000019”, “0?Return()”) in new stack
– Executing [s@sub-record-check:12] ExecIf(“SIP/1001-00000019”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [s@sub-record-check:13] GotoIf(“SIP/1001-00000019”, “0?out,1”) in new stack
– Executing [s@sub-record-check:14] Set(“SIP/1001-00000019”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:15] Set(“SIP/1001-00000019”, “NOW=1390340141”) in new stack
– Executing [s@sub-record-check:16] Set(“SIP/1001-00000019”, “__DAY=22”) in new stack
– Executing [s@sub-record-check:17] Set(“SIP/1001-00000019”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:18] Set(“SIP/1001-00000019”, “__YEAR=2014”) in new stack
– Executing [s@sub-record-check:19] Set(“SIP/1001-00000019”, “__TIMESTR=20140122-013541”) in new stack
– Executing [s@sub-record-check:20] Set(“SIP/1001-00000019”, “__FROMEXTEN=1001”) in new stack
– Executing [s@sub-record-check:21] Set(“SIP/1001-00000019”, “__CALLFILENAME=out-89261115511-1001-20140122-013541-1390340141.25”) in new stack
– Executing [s@sub-record-check:22] Goto(“SIP/1001-00000019”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] ExecIf(“SIP/1001-00000019”, “1?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [out@sub-record-check:2] GosubIf(“SIP/1001-00000019”, “0?record,1(exten,89261115511,1001)”) in new stack
– Executing [out@sub-record-check:3] Return(“SIP/1001-00000019”, “”) in new stack
– Executing [89261115511@from-internal:6] Macro(“SIP/1001-00000019”, “dialout-trunk,2,89261115511,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1001-00000019”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/1001-00000019”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/1001-00000019”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1001-00000019”, “DIAL_NUMBER=89261115511”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/1001-00000019”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/1001-00000019”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/1001-00000019”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/1001-00000019”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/1001-00000019”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/1001-00000019”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/1001-00000019”, “outbound-callerid,2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/1001-00000019”, “0?Set(REALCALLERIDNUM=1001)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/1001-00000019”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/1001-00000019”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/1001-00000019”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/1001-00000019”, “TRUNKOUTCID=1504”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/1001-00000019”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/1001-00000019”, “1?Set(CALLERID(all)=1504)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/1001-00000019”, “1?Set(CALLERID(all)=1504)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:18] Set(“SIP/1001-00000019”, “CDR(outbound_cnum)=1504”) in new stack
– Executing [s@macro-outbound-callerid:19] Set(“SIP/1001-00000019”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/1001-00000019”, “0?sub-flp-2,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/1001-00000019”, “OUTNUM=89261115511”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1001-00000019”, “custom=SIP/domspec”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/1001-00000019”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/1001-00000019”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/1001-00000019”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/1001-00000019”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/1001-00000019”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/1001-00000019”, “1?Set(CONNECTEDLINE(num,i)=89261115511)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/1001-00000019”, “1?Set(CONNECTEDLINE(name,i)=CID:1504)”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“SIP/1001-00000019”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:22] Dial(“SIP/1001-00000019”, “SIP/domspec/89261115511,300,Tt”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 16140
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK3983d828;rport
Max-Forwards: 70
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.7.0)
Date: Tue, 21 Jan 2014 21:35:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 1504026885 1504026885 IN IP4 ex.ter.nal.ip
s=Asterisk PBX 11.7.0
c=IN IP4 ex.ter.nal.ip
t=0 0
m=audio 16140 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called SIP/domspec/89261115511
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK3983d828;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as2249682e
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="60f13478"
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to tr.un.k.ip:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK3983d828;rport
Max-Forwards: 70
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as2249682e
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0
Audio is at 16140
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK539985c0;rport
Max-Forwards: 70
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.7.0)
Authorization: Digest username=“1504”, realm=“asterisk”, algorithm=MD5, uri=“sip:[email protected]”, nonce=“60f13478”, response="4ec8e6ca44b856288c40fbc575cdb6e8"
Date: Tue, 21 Jan 2014 21:35:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 1504026885 1504026886 IN IP4 ex.ter.nal.ip
s=Asterisk PBX 11.7.0
c=IN IP4 ex.ter.nal.ip
t=0 0
m=audio 16140 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
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:tr.un.k.ip:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK539985c0;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected]
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK539985c0;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as610e8394
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------->
— (11 headers 0 lines) —
list_route: hop: sip:[email protected]:5060
– SIP/domspec-0000001a is ringing
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK539985c0;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as610e8394
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 2131348254 2131348254 IN IP4 tr.un.k.ip
s=Asterisk PBX 1.8.5.0
c=IN IP4 tr.un.k.ip
t=0 0
m=audio 10216 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (12 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port tr.un.k.ip:10216
list_route: hop: sip:[email protected]:5060
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to tr.un.k.ip:5060
Transmitting (NAT) to tr.un.k.ip:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK5bd628c8;rport
Max-Forwards: 70
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as610e8394
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0
-- SIP/domspec-0000001a answered SIP/1001-00000019
Audio is at 13576
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
> 0x22f2010 – Probation passed - setting RTP source address to 192.168.127.5:10216
> 0x22f2010 – Probation passed - setting RTP source address to 192.168.127.5:10216
<— SIP read from UDP:192.168.127.5:27537 —>
OPTIONS sip:[email protected] SIP/2.0
Call-ID: [email protected]
CSeq: 1028 OPTIONS
From: “1001” sip:[email protected];tag=2449574998
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK7a707d0604afcef930df859f8fe89802323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 192.168.127.5:27537 (NAT)
Looking for 89261115511 in from-sip-external (domain ex.ter.nal.ip)
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK7a707d0604afcef930df859f8fe89802323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=2449574998
To: sip:[email protected];tag=as168980d8
Call-ID: [email protected]
CSeq: 1028 OPTIONS
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.127.6:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Retransmitting #1 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2014-01-22 01:35:49] NOTICE[7324]: chan_sip.c:15023 sip_reregister: – Re-registration for [email protected]
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
REGISTER sip:tr.un.k.ip SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK3c35d1a6;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4d8400fc
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: FPBX-2.11.0(11.7.0)
Authorization: Digest username=“1504”, realm=“asterisk”, algorithm=MD5, uri=“sip:tr.un.k.ip”, nonce=“7517c048”, response="1bad7d620f8375da655f10874699955b"
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK3c35d1a6;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as4d8400fc
To: sip:[email protected];tag=as18a0029f
Call-ID: [email protected]
CSeq: 104 REGISTER
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="50cc04d7"
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name tr.un.k.ip
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
REGISTER sip:tr.un.k.ip SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK698becb8;rport
Max-Forwards: 70
From: sip:[email protected];tag=as6a6fa714
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: FPBX-2.11.0(11.7.0)
Authorization: Digest username=“1504”, realm=“asterisk”, algorithm=MD5, uri=“sip:tr.un.k.ip”, nonce=“50cc04d7”, response="f140dd3820f9f3730c210ba0077041c2"
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0
<— SIP read from UDP:tr.un.k.ip:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP tr.un.k.ip:5060;branch=z9hG4bK18f4419f;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as67a86339
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.5.0)
Date: Tue, 21 Jan 2014 21:35:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to tr.un.k.ip:5060 (NAT)
Looking for 1504 in from-sip-external (domain ex.ter.nal.ip)
<— Transmitting (NAT) to tr.un.k.ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP tr.un.k.ip:5060;branch=z9hG4bK18f4419f;received=tr.un.k.ip;rport=5060
From: “Unknown” sip:[email protected];tag=as67a86339
To: sip:[email protected]:5060;tag=as6662bdff
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:ex.ter.nal.ip:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: OPTIONS)
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK698becb8;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as6a6fa714
To: sip:[email protected];tag=as18a0029f
Call-ID: [email protected]
CSeq: 105 REGISTER
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:[email protected]:5060;expires=120
Date: Tue, 21 Jan 2014 21:35:50 GMT
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
[2014-01-22 01:35:49] NOTICE[7324]: chan_sip.c:23472 handle_response_register: Outbound Registration: Expiry for tr.un.k.ip is 120 sec (Scheduling reregistration in 105 s)
Retransmitting #2 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Retransmitting #3 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Retransmitting #4 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<— SIP read from UDP:192.168.127.5:27537 —>
OPTIONS sip:[email protected] SIP/2.0
Call-ID: [email protected]
CSeq: 5468 OPTIONS
From: “1001” sip:[email protected];tag=515114903
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bKff1d1a8db229716e2b660a707dc941cd323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 192.168.127.5:27537 (NAT)
Looking for 89261115511 in from-sip-external (domain ex.ter.nal.ip)
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bKff1d1a8db229716e2b660a707dc941cd323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=515114903
To: sip:[email protected];tag=as40f911a0
Call-ID: [email protected]
CSeq: 5468 OPTIONS
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.127.6:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Retransmitting #5 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Retransmitting #6 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
OPTIONS sip:tr.un.k.ip SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK1fa257f6;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as21495554
To: sip:tr.un.k.ip
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Tue, 21 Jan 2014 21:36:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK1fa257f6;rport;received=ex.ter.nal.ip
From: “Unknown” sip:[email protected];tag=as21495554
To: sip:tr.un.k.ip;tag=as6ad0aa0d
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:tr.un.k.ip:5060
Accept: application/sdp
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Retransmitting #7 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<— SIP read from UDP:192.168.127.5:27537 —>
OPTIONS sip:[email protected] SIP/2.0
Call-ID: [email protected]
CSeq: 2937 OPTIONS
From: “1001” sip:[email protected];tag=1434849017
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bKe14f4c35d43c99707f2555d2ed3fcee1323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 192.168.127.5:27537 (NAT)
Looking for 89261115511 in from-sip-external (domain ex.ter.nal.ip)
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bKe14f4c35d43c99707f2555d2ed3fcee1323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=1434849017
To: sip:[email protected];tag=as6d6a7ee7
Call-ID: [email protected]
CSeq: 2937 OPTIONS
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.127.6:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
Retransmitting #8 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Really destroying SIP dialog ‘[email protected]’ Method: ACK
Retransmitting #9 (NAT) to 192.168.127.5:27537:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK252579f1a12dccdb3b6058016d26dd16323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=905239798
To: sip:[email protected];tag=as770f8533
Call-ID: [email protected]
CSeq: 6871 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 59442835 59442835 IN IP4 192.168.127.6
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.127.6
t=0 0
m=audio 13576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<— SIP read from UDP:192.168.127.5:27537 —>
OPTIONS sip:[email protected] SIP/2.0
Call-ID: [email protected]
CSeq: 2596 OPTIONS
From: “1001” sip:[email protected];tag=3645356788
To: sip:[email protected]
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK260738d0846b915f9ea699fcd7ea05d4323437;rport
Max-Forwards: 70
Contact: “1001” sip:[email protected]:33574;transport=udp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 192.168.127.5:27537 (NAT)
Looking for 89261115511 in from-sip-external (domain ex.ter.nal.ip)
<— Transmitting (NAT) to 192.168.127.5:27537 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:33574;branch=z9hG4bK260738d0846b915f9ea699fcd7ea05d4323437;received=192.168.127.5;rport=27537
From: “1001” sip:[email protected];tag=3645356788
To: sip:[email protected];tag=as6d3c757e
Call-ID: [email protected]
CSeq: 2596 OPTIONS
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.127.6:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
[2014-01-22 01:36:19] NOTICE[7324]: chan_sip.c:28883 check_rtp_timeout: Disconnecting call ‘SIP/1001-00000019’ for lack of RTP activity in 31 seconds
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/1001-00000019”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/1001-00000019”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/1001-00000019”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/1001-00000019”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1001-00000019’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/1001-00000019’
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to tr.un.k.ip:5060
Reliably Transmitting (NAT) to tr.un.k.ip:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK033604f8;rport
Max-Forwards: 70
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as610e8394
Call-ID: [email protected]:5060
CSeq: 104 BYE
User-Agent: FPBX-2.11.0(11.7.0)
Authorization: Digest username=“1504”, realm=“asterisk”, algorithm=MD5, uri=“sip:[email protected]:5060”, nonce=“60f13478”, response="3e06f1cf61cde5183023cbe4eb4c383d"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/1001-00000019’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 89261115511, 6) exited non-zero on 'SIP/1001-00000019’
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:tr.un.k.ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ex.ter.nal.ip:5060;branch=z9hG4bK033604f8;received=ex.ter.nal.ip;rport=5060
From: sip:[email protected];tag=as11ae80e7
To: sip:[email protected];tag=as610e8394
Call-ID: [email protected]:5060
CSeq: 104 BYE
Server: FPBX-2.10.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
[2014-01-22 01:36:20] WARNING[7324]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 6871 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32108ms with no response
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
*CLI> sip set debug off