Error SIP/2.0 401 Unauthorized and SIP/2.0 503 Service Unava

Hi there
Please, I need some help. I can’t make an outgoing/incoming calls. I searched in another Forums but this error caused by different issues. I changed some characters with xx by security reason. Thanks by your help.

INVITE sip:XXX3687661@xx.xx.204.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 73.221.112.174:57713;rport;branch=z9hG4bKPj6zz-GABHsqW8xK2JK2ff79jmCzfkAR6w
Max-Forwards: 70
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2
Contact: 189" sip:189@73.221.112.174:57713;+sip.ice
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26451 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, MESSAGE
Supported: replaces, 100rel, timer, norefersub, eventlist
User-Agent: Join 3.2.2(8905)
Content-Type: application/sdp
Content-Length: 676

v=0
o=- 3630853166 3630853166 IN IP4 192.168.0.103
s=v
b=AS:241
t=0 0
m=audio 1028 RTP/AVP 0 8 3 9 107 105 109 102 125 101
c=IN IP4 73.221.112.174
b=TIAS:64000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 SILK/16000
a=rtpmap:105 SILK/8000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:102 speex/8000
a=rtpmap:125 opus/48000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 1030 RTP/AVP 115 34 121
c=IN IP4 73.221.112.174
b=TIAS:150000
a=sendrecv
a=rtpmap:115 H263-1998/90000
a=fmtp:115 CIF=1;QCIF=1
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=1;QCIF=1
a=rtpmap:121 VP8/90000
<------------->
— (13 headers 30 lines) —
Sending to 73.221.112.174:57713 (no NAT)
Sending to 73.221.112.174:57713 (no NAT)
Using INVITE request as basis request - vopxjovDG-wGqBakMs1aaLNMSOa90no4
Found peer ‘189’ for ‘189’ from 73.221.112.174:57713

<— Reliably Transmitting (no NAT) to 73.221.112.174:57713 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 73.221.112.174:57713;branch=z9hG4bKPj6zz-GABHsqW8xK2JK2ff79jmCzfkAR6w;received=73.221.112.174;rport=57713
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2;tag=as121cb744
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26451 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7cb3e2e4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘vopxjovDG-wGqBakMs1aaLNMSOa90no4’ in 6848 ms (Method: INVITE)

<— SIP read from UDP:73.221.112.174:57713 —>
ACK sip:XXX3687661@xx.xx.204.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 73.221.112.174:57713;rport;branch=z9hG4bKPj6zz-GABHsqW8xK2JK2ff79jmCzfkAR6w
Max-Forwards: 70
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2;tag=as121cb744
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26451 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:73.221.112.174:57713 —>
INVITE sip:XXX3687661@xx.xx.204.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 73.221.112.174:57713;rport;branch=z9hG4bKPjhEIRK2JtzyU5CXx7ZAORfeBhW9CFdhtI
Max-Forwards: 70
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2
Contact: “189” sip:189@73.221.112.174:57713;+sip.ice
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26452 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, MESSAGE
Supported: replaces, 100rel, timer, norefersub, eventlist
User-Agent: Join 3.2.2(8905)
Authorization: Digest username=“189”, realm=“asterisk”, nonce=“7cb3e2e4”, uri=“sip:XXX3687661@xx.xx.204.2;transport=udp”, response=“508d5a4aee35f11cbf7865f446fe4a6f”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 676

v=0
o=- 3630853166 3630853166 IN IP4 192.168.0.103
s=v
b=AS:241
t=0 0
m=audio 1028 RTP/AVP 0 8 3 9 107 105 109 102 125 101
c=IN IP4 73.221.112.174
b=TIAS:64000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 SILK/16000
a=rtpmap:105 SILK/8000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:102 speex/8000
a=rtpmap:125 opus/48000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 1030 RTP/AVP 115 34 121
c=IN IP4 73.221.112.174
b=TIAS:150000
a=sendrecv
a=rtpmap:115 H263-1998/90000
a=fmtp:115 CIF=1;QCIF=1
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=1;QCIF=1
a=rtpmap:121 VP8/90000
<------------->
— (14 headers 30 lines) —
Sending to 73.221.112.174:57713 (no NAT)
Using INVITE request as basis request - vopxjovDG-wGqBakMs1aaLNMSOa90no4
Found peer ‘189’ for ‘189’ from 73.221.112.174:57713
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 107
Found RTP audio format 105
Found RTP audio format 109
Found RTP audio format 102
Found RTP audio format 125
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 GSM for ID 3
Found audio description format G722 for ID 9
Found audio description format SILK for ID 107
Found audio description format SILK for ID 105
Found audio description format iLBC for ID 109
Found audio description format speex for ID 102
Found unknown media description format opus for ID 125
Found audio description format telephone-event for ID 101
Found RTP video format 115
Found RTP video format 34
Found RTP video format 121
Found video description format H263-1998 for ID 115
Found video description format H263 for ID 34
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc|g722|silk8|silk16)/video=(h263|h263p)/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 73.221.112.174:1028
Looking for XXX3687661 in from-internal (domain xx.xx.204.2)
list_route: hop: sip:189@73.221.112.174:57713

<— Transmitting (no NAT) to 73.221.112.174:57713 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 73.221.112.174:57713;branch=z9hG4bKPjhEIRK2JtzyU5CXx7ZAORfeBhW9CFdhtI;received=73.221.112.174;rport=57713
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26452 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:XXX3687661@xx.xx.204.2:5060
Content-Length: 0

<------------>
– Executing [XXX3687661@from-internal:1] Macro(“SIP/189-00000067”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/189-00000067”, “TOUCH_MONITOR=1421864365.103”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/189-00000067”, “AMPUSER=189”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/189-00000067”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/189-00000067”, “1?Set(REALCALLERIDNUM=189)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/189-00000067”, “AMPUSER=189”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/189-00000067”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/189-00000067”, “AMPUSERCIDNAME=189”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/189-00000067”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/189-00000067”, “AMPUSERCID=189”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/189-00000067”, “__DIAL_OPTIONS=tr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/189-00000067”, “CALLERID(all)=“189” <189>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/189-00000067”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/189-00000067”, “1?Set(GROUP(concurrency_limit)=189)”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/189-00000067”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/189-00000067”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,28)
– Executing [s@macro-user-callerid:28] Set(“SIP/189-00000067”, “CALLERID(number)=189”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/189-00000067”, “CALLERID(name)=189”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/189-00000067”, “CDR(cnum)=189”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/189-00000067”, “CDR(cnam)=189”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/189-00000067”, “CHANNEL(language)=en”) in new stack
– Executing [XXX3687661@from-internal:2] Set(“SIP/189-00000067”, “MOHCLASS=default”) in new stack
– Executing [XXX3687661@from-internal:3] Set(“SIP/189-00000067”, “_NODEST=”) in new stack
– Executing [XXX3687661@from-internal:4] Gosub(“SIP/189-00000067”, “sub-record-check,s,1(out,XXX3687661,)”) in new stack
– Executing [s@sub-record-check:1] Set(“SIP/189-00000067”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:2] GotoIf(“SIP/189-00000067”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [s@sub-record-check:7] Set(“SIP/189-00000067”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:8] GotoIf(“SIP/189-00000067”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [s@sub-record-check:11] ExecIf(“SIP/189-00000067”, “0?Return()”) in new stack
– Executing [s@sub-record-check:12] ExecIf(“SIP/189-00000067”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [s@sub-record-check:13] GotoIf(“SIP/189-00000067”, “0?out,1”) in new stack
– Executing [s@sub-record-check:14] Set(“SIP/189-00000067”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:15] Set(“SIP/189-00000067”, “NOW=1421864365”) in new stack
– Executing [s@sub-record-check:16] Set(“SIP/189-00000067”, “__DAY=21”) in new stack
– Executing [s@sub-record-check:17] Set(“SIP/189-00000067”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:18] Set(“SIP/189-00000067”, “__YEAR=2015”) in new stack
– Executing [s@sub-record-check:19] Set(“SIP/189-00000067”, “__TIMESTR=20150121-131925”) in new stack
– Executing [s@sub-record-check:20] Set(“SIP/189-00000067”, “__FROMEXTEN=189”) in new stack
– Executing [s@sub-record-check:21] Set(“SIP/189-00000067”, “__CALLFILENAME=out-XXX3687661-189-20150121-131925-1421864365.103”) in new stack
– Executing [s@sub-record-check:22] Goto(“SIP/189-00000067”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] ExecIf(“SIP/189-00000067”, “1?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [out@sub-record-check:2] GosubIf(“SIP/189-00000067”, “0?record,1(exten,XXX3687661,189)”) in new stack
– Executing [out@sub-record-check:3] Return(“SIP/189-00000067”, “”) in new stack
– Executing [XXX3687661@from-internal:5] Macro(“SIP/189-00000067”, “dialout-trunk,2,XXX3687661,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/189-00000067”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/189-00000067”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/189-00000067”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/189-00000067”, “DIAL_NUMBER=XXX3687661”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/189-00000067”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/189-00000067”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/189-00000067”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/189-00000067”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/189-00000067”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/189-00000067”, “outbound-callerid,2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/189-00000067”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/189-00000067”, “0?Set(REALCALLERIDNUM=189)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/189-00000067”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/189-00000067”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/189-00000067”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/189-00000067”, “TRUNKOUTCID=7865422691”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/189-00000067”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/189-00000067”, “1?Set(CALLERID(all)=7865422691)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/189-00000067”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/189-00000067”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/189-00000067”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:18] Set(“SIP/189-00000067”, “CDR(outbound_cnum)=7865422691”) in new stack
– Executing [s@macro-outbound-callerid:19] Set(“SIP/189-00000067”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/189-00000067”, “0?sub-flp-2,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/189-00000067”, “OUTNUM=XXX3687661”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/189-00000067”, “custom=SIP/winds”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/189-00000067”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/189-00000067”, “0?Set(DIAL_TRUNK_OPTIONS=M(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/189-00000067”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/189-00000067”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/189-00000067”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/189-00000067”, “1?Set(CONNECTEDLINE(num,i)=XXX3687661)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/189-00000067”, “1?Set(CONNECTEDLINE(name,i)=CID:7865422691)”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“SIP/189-00000067”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:22] Dial(“SIP/189-00000067”, “SIP/winds/XXX3687661,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 11056
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.250.0.5:5060:
INVITE sip:XXX3687661@10.250.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK298a9e3f;rport
Max-Forwards: 70
From: sip:XXX5422711@=10.250.0.X;tag=as09035e0b
To: sip:XXX3687661@10.250.0.X:5060
Contact: sip:XXX5422711@192.168.0.5:5060
Call-ID: 46e810056a6b639d4f2ece34206b180c@=10.250.0.X
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 21 Jan 2015 18:19:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 328

v=0
o=root 1913029695 1913029695 IN IP4 192.168.0.5
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.0.5
t=0 0
m=audio 11056 RTP/AVP 8 18 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/winds/XXX3687661

<— SIP read from UDP:10.250.0.X:5060 —>
SIP/2.0 400 Invalid From
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK298a9e3f;rport=5060
From: sip:XXX5422711@=10.250.0.X;tag=as09035e0b
To: sip:XXX3687661@10.250.0.X:5060;tag=aprqngfrt-u8184830000c6
Call-ID: 46e810056a6b639d4f2ece34206b180c@=10.250.0.X
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —
– Got SIP response 400 “Invalid From” back from 10.250.0.X:5060
Transmitting (NAT) to 10.250.0.X:5060:
ACK sip:XXX3687661@10.250.0.X:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK298a9e3f;rport
Max-Forwards: 70
From: sip:XXX5422711@=10.250.0.5;tag=as09035e0b
To: sip:XXX3687661@10.250.0.5:5060;tag=aprqngfrt-u8184830000c6
Contact: sip:XXX5422711@192.168.0.5:5060
Call-ID: 46e810056a6b639d4f2ece34206b180c@=10.250.0.X
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


-- SIP/winds-00000068 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:23] NoOp(“SIP/189-00000067”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 127”) in new stack
– Executing [s@macro-dialout-trunk:24] GotoIf(“SIP/189-00000067”, “0?continue,1:s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/189-00000067”, “RC=127”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/189-00000067”, “127,1”) in new stack
– Goto (macro-dialout-trunk,127,1)
– Executing [127@macro-dialout-trunk:1] Goto(“SIP/189-00000067”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/189-00000067”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 127 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] Set(“SIP/189-00000067”, “CALLERID(number)=189”) in new stack
– Executing [XXX3687661@from-internal:6] Macro(“SIP/189-00000067”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/189-00000067”, “”) in new stack
Audio is at 18348
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 73.221.112.174:57713 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 73.221.112.174:57713;branch=z9hG4bKPjhEIRK2JtzyU5CXx7ZAORfeBhW9CFdhtI;received=73.221.112.174;rport=57713
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2;tag=as1552b8d0
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26452 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:XXX3687661@xx.xx.204.2:5060
Content-Type: application/sdp
Require: timer
Content-Length: 307

v=0
o=root 350423923 350423923 IN IP4 xx.xx.204.2
s=Asterisk PBX 11.13.0
c=IN IP4 xx.xx.204.2
t=0 0
m=audio 18348 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 115 34 121

<------------>
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/189-00000067”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/189-00000067”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/189-00000067”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/189-00000067> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
Reliably Transmitting (no NAT) to 73.221.112.174:57713:
OPTIONS sip:189@73.221.112.174:57713 SIP/2.0
Via: SIP/2.0/UDP xx.xx.204.2:5060;branch=z9hG4bK47691de8
Max-Forwards: 70
From: “Unknown” sip:Unknown@xx.xx.204.2;tag=as47d9e1cb
To: sip:189@73.221.112.174:57713
Contact: sip:Unknown@xx.xx.204.2:5060
Call-ID: 2bc892094a838af24ca04e007b270cdf@xx.xx.204.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 21 Jan 2015 18:19:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog ‘46e810056a6b639d4f2ece34206b180c@=10.250.0.5’ Method: INVITE
> 0x2b39d81f5630 – Probation passed - setting RTP source address to 73.221.112.174:1028

<— SIP read from UDP:73.221.112.174:57713 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.9.204.2:5060;received=74.9.204.2;branch=z9hG4bK47691de8
Call-ID: 2bc892094a838af24ca04e007b270cdf@74.9.204.2:5060
From: “Unknown” sip:Unknown@xx.xx.204.2;tag=as47d9e1cb
To: sip:189@73.221.112.174:1026;tag=z9hG4bK47691de8
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, MESSAGE
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/rlmi+xml, multipart/related, application/watcherinfo+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain, text/plain
Supported: replaces, 100rel, timer, norefersub, eventlist
Allow-Events: presence, presence.winfo, message-summary, refer
User-Agent: Join 3.2.2(8905)
Contact: “189” sip:189@73.221.112.174:57713;+sip.ice
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog '2bc892094a838af24ca04e007b270cdf@xx.xx.204.2:5060’ Method: OPTIONS
– <SIP/189-00000067> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“SIP/189-00000067”, “20”) in new stack

<— Reliably Transmitting (no NAT) to 73.221.112.174:57713 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 73.221.112.174:57713;branch=z9hG4bKPjhEIRK2JtzyU5CXx7ZAORfeBhW9CFdhtI;received=73.221.112.174;rport=57713
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2;tag=as1552b8d0
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26452 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0

<------------>
[2015-01-21 13:19:29] WARNING[17197][C-0000004d]: channel.c:4860 ast_prod: Prodding channel ‘SIP/189-00000067’ failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/189-00000067’ in macro ‘outisbusy’
== Spawn extension (from-internal, XXX3687661, 6) exited non-zero on ‘SIP/189-00000067’
– Executing [h@from-internal:1] Hangup(“SIP/189-00000067”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/189-00000067’

<— SIP read from UDP:73.221.112.174:57713 —>
ACK sip:XXX3687661@xx.xx.204.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 73.221.112.174:57713;rport;branch=z9hG4bKPjhEIRK2JtzyU5CXx7ZAORfeBhW9CFdhtI
Max-Forwards: 70
From: “189” sip:189@xx.xx.204.2;tag=nX6eJJAn8qPlQLWhHGDoqjv.eqmy52mo
To: sip:XXX3687661@xx.xx.204.2;tag=as1552b8d0
Call-ID: vopxjovDG-wGqBakMs1aaLNMSOa90no4
CSeq: 26452 ACK
Content-Length: 0

You should contact the FreePBX community on community.freepbx.org/

However, it looks to me as though there is an error in you fromdomain in sip.conf. Specifically, I think you have fromdomain==

I concur. The other end tells you this:

<--- SIP read from UDP:10.250.0.X:5060 --->
SIP/2.0 400 Invalid From
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK298a9e3f;rport=5060
From: <sip:XXX5422711@=10.250.0.X>;tag=as09035e0b
To: <sip:XXX3687661@10.250.0.X:5060>;tag=aprqngfrt-u8184830000c6
Call-ID: 46e810056a6b639d4f2ece34206b180c@=10.250.0.X
CSeq: 102 INVITE

[b]400 Bad Request

The request could not be understood due to malformed syntax. The
Reason-Phrase SHOULD identify the syntax problem in more detail, for
example, “Missing Call-ID header field”[/b]

check your sip configuration for that peer