FreePBX NAT

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: 4b741f90de4d6f274e7a2528f5c409d3[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 [[email protected]:1] Macro(“SIP/1001-00000019”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/1001-00000019”, “TOUCH_MONITOR=1390340141.25”) in new stack
– Executing [[email protected]:2] Set(“SIP/1001-00000019”, “AMPUSER=1001”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1001-00000019”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/1001-00000019”, “1?Set(REALCALLERIDNUM=1001)”) in new stack
– Executing [[email protected]:5] Set(“SIP/1001-00000019”, “AMPUSER=1001”) in new stack
– Executing [[email protected]:6] Set(“SIP/1001-00000019”, “AMPUSERCIDNAME=Alex”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/1001-00000019”, “0?report”) in new stack
– Executing [[email protected]:8] Set(“SIP/1001-00000019”, “AMPUSERCID=1001”) in new stack
– Executing [[email protected]:9] Set(“SIP/1001-00000019”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:10] Set(“SIP/1001-00000019”, “CALLERID(all)=“Alex” <1001>”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/1001-00000019”, “0?limit”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/1001-00000019”, “1?Set(GROUP(concurrency_limit)=1001)”) in new stack
– Executing [[email protected]:13] GosubIf(“SIP/1001-00000019”, “7?sub-ccss,s,1(from-internal,89261115511)”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/1001-00000019”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/1001-00000019”, “CCSS_SETUP=TRUE”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/1001-00000019”, “0?monitor_config,1(from-internal,89261115511):monitor_default,1(from-internal,89261115511)”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1001-00000019”, “0?is_exten”) in new stack
– Executing [m[email protected]:2] StackPop(“SIP/1001-00000019”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/1001-00000019”, “FALSE”) in new stack
– Executing [[email protected]:14] GotoIf(“SIP/1001-00000019”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,27)
– Executing [[email protected]:27] Set(“SIP/1001-00000019”, “CALLERID(number)=1001”) in new stack
– Executing [[email protected]:28] Set(“SIP/1001-00000019”, “CALLERID(name)=Alex”) in new stack
– Executing [[email protected]:29] Set(“SIP/1001-00000019”, “CDR(cnum)=1001”) in new stack
– Executing [[email protected]:30] Set(“SIP/1001-00000019”, “CDR(cnam)=Alex”) in new stack
– Executing [[email protected]:31] Set(“SIP/1001-00000019”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Set(“SIP/1001-00000019”, “MOHCLASS=default”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/1001-00000019”, “1?Set(TRUNKCIDOVERRIDE=1504)”) in new stack
– Executing [[email protected]:4] Set(“SIP/1001-00000019”, “_NODEST=”) in new stack
– Executing [[email protected]:5] Gosub(“SIP/1001-00000019”, “sub-record-check,s,1(out,89261115511,)”) in new stack
– Executing [[email protected]:1] Set(“SIP/1001-00000019”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/1001-00000019”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [[email protected]:7] Set(“SIP/1001-00000019”, “__MON_FMT=wav”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/1001-00000019”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [[email protected]:11] ExecIf(“SIP/1001-00000019”, “0?Return()”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/1001-00000019”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/1001-00000019”, “0?out,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/1001-00000019”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [[email protected]:15] Set(“SIP/1001-00000019”, “NOW=1390340141”) in new stack
– Executing [[email protected]:16] Set(“SIP/1001-00000019”, “__DAY=22”) in new stack
– Executing [[email protected]:17] Set(“SIP/1001-00000019”, “__MONTH=01”) in new stack
– Executing [[email protected]:18] Set(“SIP/1001-00000019”, “__YEAR=2014”) in new stack
– Executing [[email protected]:19] Set(“SIP/1001-00000019”, “__TIMESTR=20140122-013541”) in new stack
– Executing [[email protected]:20] Set(“SIP/1001-00000019”, “__FROMEXTEN=1001”) in new stack
– Executing [[email protected]:21] Set(“SIP/1001-00000019”, “__CALLFILENAME=out-89261115511-1001-20140122-013541-1390340141.25”) in new stack
– Executing [[email protected]:22] Goto(“SIP/1001-00000019”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [[email protected]:1] ExecIf(“SIP/1001-00000019”, “1?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/1001-00000019”, “0?record,1(exten,89261115511,1001)”) in new stack
– Executing [[email protected]:3] Return(“SIP/1001-00000019”, “”) in new stack
– Executing [[email protected]:6] Macro(“SIP/1001-00000019”, “dialout-trunk,2,89261115511,off”) in new stack
– Executing [[email protected]:1] Set(“SIP/1001-00000019”, “DIAL_TRUNK=2”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/1001-00000019”, “0?sub-pincheck,s,1()”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1001-00000019”, “0?disabletrunk,1”) in new stack
– Executing [[email protected]o-dialout-trunk:4] Set(“SIP/1001-00000019”, “DIAL_NUMBER=89261115511”) in new stack
– Executing [[email protected]:5] Set(“SIP/1001-00000019”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:6] Set(“SIP/1001-00000019”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/1001-00000019”, “0?nomax”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/1001-00000019”, “0?chanfull”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/1001-00000019”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/1001-00000019”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [[email protected]:11] Macro(“SIP/1001-00000019”, “outbound-callerid,2”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/1001-00000019”, “0?Set(REALCALLERIDNUM=1001)”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1001-00000019”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/1001-00000019”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/1001-00000019”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/1001-00000019”, “TRUNKOUTCID=1504”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/1001-00000019”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [[email protected]:14] ExecIf(“SIP/1001-00000019”, “1?Set(CALLERID(all)=1504)”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/1001-00000019”, “1?Set(CALLERID(all)=1504)”) in new stack
– Executing [[email protected]:17] ExecIf(“SIP/1001-00000019”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [[email protected]:18] Set(“SIP/1001-00000019”, “CDR(outbound_cnum)=1504”) in new stack
– Executing [[email protected]:19] Set(“SIP/1001-00000019”, “CDR(outbound_cnam)=”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/1001-00000019”, “0?sub-flp-2,s,1()”) in new stack
– Executing [[email protected]:13] Set(“SIP/1001-00000019”, “OUTNUM=89261115511”) in new stack
– Executing [[email protected]:14] Set(“SIP/1001-00000019”, “custom=SIP/domspec”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/1001-00000019”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/1001-00000019”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [[email protected]:17] Macro(“SIP/1001-00000019”, “dialout-trunk-predial-hook,”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/1001-00000019”, “”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/1001-00000019”, “0?bypass,1”) in new stack
– Executing [[email protected]:19] ExecIf(“SIP/1001-00000019”, “1?Set(CONNECTEDLINE(num,i)=89261115511)”) in new stack
– Executing [[email protected]:20] ExecIf(“SIP/1001-00000019”, “1?Set(CONNECTEDLINE(name,i)=CID:1504)”) in new stack
– Executing [[email protected]:21] GotoIf(“SIP/1001-00000019”, “0?customtrunk”) in new stack
– Executing [[email protected]: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 ‘1ffc17ad6c7[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 [[email protected]:1] Macro(“SIP/1001-00000019”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/1001-00000019”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] ExecIf(“SIP/1001-00000019”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]: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

You don’t have the settings under NAT in SIP settings set properly

Excuse me, i didn’t get it.
Where is the problem? which file?

SIP Settings FreePBX module, NAT section, right on top.

Everything is fine there.
Nat: yes
IP Configuration: static
External IP: 1xx.xxx.1xx.2xx
Local Networks: 192.168.127.4/255.255.255.252

May be the problem is in asterisk? i just installed the latest version 11.7.0
when I make
*CLI> module reload chan_sip.so

== Parsing ‘/etc/asterisk/sip_additional.conf’: Found
== Parsing ‘/etc/asterisk/sip_custom_post.conf’: Found
[2014-01-23 00:58:39] WARNING[14154]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead

Still can’t make it. Patched asterisk with https://issues.asterisk.org/jira/browse/ASTERISK-20674 and https://issues.asterisk.org/jira/browse/ASTERISK-21225
Doesn’t help…

All settings are fine…

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.11.0(11.7.0)
SDP Session Name: Asterisk PBX 11.7.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: my.ex.t.ip:0
Externrefresh: 10
Localnet: 192.168.127.4/255.255.255.252

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|speex)
Codec Order: ulaw:20,alaw:20,gsm:20,speex:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 900 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:

But still receive:
Sent RTP packet to 192.168.1.108:56220 (type 00, seq 047771, ts 1367808592, len 000160)
Got RTP packet from 192.168.127.5:10234 (type 00, seq 055474, ts 1367808752, len 000160)

If the phones are in the in the 192.168.1.0/24 network you are not NAT’ing to them you are routing to them Add 192.168.1.0/255.255.255.0 to the localnet

But this network is not connected to PBX server. Both phones and PBX server are NAT’ing.
192.168.1.0/24(phones local IP) - router - internet - router - 192.168.127.4/32(PBX local IP)

And somehow packets are send to and from the network that is not connected to PBX server and are dropped by router when PBX answers to that address.

Then the far end is writing the wrong address into the SIP header. Do you have some kind of SIP ALG or helper on the remote router?

localnet is a poorly named term in Asterisk. Localnets are networks that are to be excluded from NAT processing. Any network that is reachable from the adjacent gateway without NAT or other type of IP over subscription need to be excluded. Example include VPN’s or MPLS. It seems to me that you might have a tunnel or a VPN between these two locations or the above SIP helper issue.

I don’t have some strange configuration of my router. I tried to register under same extension from another PC with X-lite, from another PC and network and from 3g. Still the same, RTP packets are sent to local ip of my extension (192.168.1.0/24, 192.168.88.0/24(another network), 10.47.52.158(3g)).

I understand that there must be my external IP address in packets headers, but somehow it isn’t. I guess that asterisk doesn’t read IP address from headers, may be asterisk reads data from UDP packet’s DATA field, and asterisk asks to provide IP address of connected extension? And phones sends local IP address? This is the only reason why my phone’s local IP address is available for asterisk through internet. And the conclusion to me is to make configuration for asterisk do not ask extension for ip address and use IP address from UDP header. But i don’t know how to do it.

Very strange problem, i have configured asterisk 10.X on another server under NAT and everything works great there. I compared configuration and debug with it and i can’t see the main difference. May be I should open debug ticket on asterisk forums?

I made a tcpdump portrange 10500-20000 or port 5060 and i can see that SIP registration packets are NAT’ing well, but RTP are send to local address, but no packets with local address in header was received before. The problem is just in RTP.

Any ideas?

Spend a lot of time to fix the issue…
I re-installed asterisk, then OS + asterisk, than installed FreePBX distro Stable.

The same error every time.
found this BUG report with the same error, but closed in 2005.
My phone doesn’t have Stun server in options, “externIP= in sip.conf” is written and I don’t understand what that mean: “other techniques (rport) to discover the outside address and tell the other end about it”

Can you explain your design philosophy behind putting a single homed asterisk server in it’s own /30 network, to me it just seems a pointless exercise. Why not use a /32 network :wink: and get that complication out of the way?

Why not use a /32 network
O_o

As I wrote before, I have a single server, with 1 external IP and multiple VirtualBOX containers with different OS installed. Each container is isolated from another and they don’t need more than /30 network mask.

I installed linphone on my notebook, configured STUN server and firured out that asterisk server still send SDP information to extension to use 192.168.127.14 IP. But I HAVE exteranl IP configured in sip.conf!

sip show settings

Global Settings:

UDP Bindaddress: 192.168.127.14:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.11.0(11.7.0)
SDP Session Name: Asterisk PBX 11.7.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: 1xx.xxx.1xx.2xx:0
Externrefresh: 10
Localnet: 192.168.127.12/255.255.255.252

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|speex|g722)
Codec Order: ulaw:20,alaw:20,speex:20,g722:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 300
RTP Timeout: 60
RTP Hold Timeout: 900
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

Enabled STUN server on linphone and in iptables on my notebook:

iptables -t nat -I OUTPUT --dest 192.168.127.14 -j DNAT --to-dest ex.t.ip.add

and I can hear!!! But i can’t use this, just for tests. What is wrong?

i believe for your case your /30 networks are four times larger than they need be double and treble natting needs to be routed perhaps better than the basic nat that vbox provides. But knock yourself out with your way :slight_smile:

Please insert this option in FreePBX:

; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
;
; media_address = 172.16.42.1

This was the problem.

Found another new problem with NAT.
Option externIP in sip_general_additional.conf (included file in sip.conf general section) doesn’t works as expected. “sip show settings” looks right but “Contact” field of SDP packages contain wrong IP address.

How to fix:
Include an option “externADDR=” in sip.conf general section directly because this option doesn’t works from included file.

Please fix this issue or write about it somewhere in wiki.

Found in:
FreePBX DISTRO - Stable-5.211.65-4 Release Date-01-08-13 FreePBX 2.11, Centos 6.4
Asterisk 11. (FreePBX 2.11.0, Asterisk 11.7.0)
Manual installation: CentOS 6.5 or Debian 7 + FreePBX 2.11.0 + Asterisk 11.7.0