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"Asterisk 11 - регистрация транка"
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"Asterisk 11 - регистрация транка"  +/
Сообщение от suslic (ok) on 13-Ноя-14, 13:09 
Проблема в том, што прописавши параметры транка в SIP телефон - все хорошо. Но когда пытаюсь зделать транком на астериске - получаю Forbidden


Лог дзвонка из номера 123123 в астериске через транк 380892500614@sip.ukrtel.net на 0324771136

Спасибо.

-----------------------------------------------------
<--- SIP read from UDP:192.168.120.134:5060 --->
INVITE sip:0324771136@192.168.120.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83b373339fb1f74;rport
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 13 INVITE
Contact: <sip:123123@192.168.120.134:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
P-Early-Media: supported
User-Agent: SIPPER for phoner
P-Preferred-Identity: <sip:380892500614@192.168.120.21>
Content-Length: 423

v=0
o=- 4090330305 1 IN IP4 192.168.120.134
s=SIPPER for phoner
c=IN IP4 192.168.120.134
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 111 112 113 107
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:107 opus/48000/2
a=ssrc:117280013
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.120.134:5060 (NAT)
Sending to 192.168.120.134:5060 (NAT)
Using INVITE request as basis request - 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
Found peer '123123' for '123123' from 192.168.120.134:5060

<--- Reliably Transmitting (NAT) to 192.168.120.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83b373339fb1f74;received=192.168.120.134;rport=5060
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as07f811bd
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 13 INVITE
Server: FPBX-2.11.0(11.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62614bf6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.120.134:5060 --->
ACK sip:0324771136@192.168.120.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83b373339fb1f74;rport
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as07f811bd
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 13 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.120.134:5060 --->
INVITE sip:0324771136@192.168.120.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;rport
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 INVITE
Contact: <sip:123123@192.168.120.134:5060>
Authorization: Digest username="123123", realm="asterisk", nonce="62614bf6", uri="sip:0324771136@192.168.120.21", response="6c1d50f17a25fd62a5696c8be2ca37e9", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
P-Early-Media: supported
User-Agent: SIPPER for phoner
P-Preferred-Identity: <sip:380892500614@192.168.120.21>
Content-Length: 423

v=0
o=- 4090330305 1 IN IP4 192.168.120.134
s=SIPPER for phoner
c=IN IP4 192.168.120.134
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 111 112 113 107
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:107 opus/48000/2
a=ssrc:117280013
a=sendrecv
<------------->
--- (16 headers 18 lines) ---
Sending to 192.168.120.134:5060 (NAT)
Using INVITE request as basis request - 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
Found peer '123123' for '123123' from 192.168.120.134:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format speex for ID 111
Found unknown media description format G726-16 for ID 112
Found unknown media description format G726-24 for ID 113
Found unknown media description format opus for ID 107
Capabilities: us - (gsm|ulaw|alaw|g726|g729), peer - audio=(gsm|ulaw|alaw|g726|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.120.134:5062
Looking for 0324771136 in from-internal (domain 192.168.120.21)
list_route: hop: <sip:123123@192.168.120.134:5060>

<--- Transmitting (NAT) to 192.168.120.134:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;received=192.168.120.134;rport=5060
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 INVITE
Server: FPBX-2.11.0(11.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0324771136@192.168.120.21:5060>
Content-Length: 0


<------------>
    -- Executing [0324771136@from-internal:1] Macro("SIP/123123-00000008", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/123123-00000008", "TOUCH_MONITOR=1415873130.8") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/123123-00000008", "AMPUSER=123123") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/123123-00000008", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/123123-00000008", "1?Set(REALCALLERIDNUM=123123)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/123123-00000008", "AMPUSER=123123") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/123123-00000008", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/123123-00000008", "AMPUSERCIDNAME=123123") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/123123-00000008", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/123123-00000008", "AMPUSERCID=123123") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/123123-00000008", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/123123-00000008", "CALLERID(all)="123123" <123123>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/123123-00000008", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/123123-00000008", "1?Set(GROUP(concurrency_limit)=123123)") in new stack
    -- Executing [s@macro-user-callerid:14] GosubIf("SIP/123123-00000008", "7?sub-ccss,s,1(from-internal,0324771136)") in new stack
    -- Executing [s@sub-ccss:1] ExecIf("SIP/123123-00000008", "0?Return()") in new stack
    -- Executing [s@sub-ccss:2] Set("SIP/123123-00000008", "CCSS_SETUP=TRUE") in new stack
    -- Executing [s@sub-ccss:3] GosubIf("SIP/123123-00000008", "0?monitor_config,1(from-internal,0324771136):monitor_default,1(from-internal,0324771136)") in new stack
    -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/123123-00000008", "0?is_exten") in new stack
    -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/123123-00000008", "") in new stack
    -- Executing [monitor_default@sub-ccss:3] Return("SIP/123123-00000008", "FALSE") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("SIP/123123-00000008", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,28)
    -- Executing [s@macro-user-callerid:28] Set("SIP/123123-00000008", "CALLERID(number)=123123") in new stack
    -- Executing [s@macro-user-callerid:29] Set("SIP/123123-00000008", "CALLERID(name)=123123") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/123123-00000008", "CDR(cnum)=123123") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/123123-00000008", "CDR(cnam)=123123") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/123123-00000008", "CHANNEL(language)=en") in new stack
    -- Executing [0324771136@from-internal:2] Set("SIP/123123-00000008", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [0324771136@from-internal:3] Set("SIP/123123-00000008", "MOHCLASS=default") in new stack
    -- Executing [0324771136@from-internal:4] Set("SIP/123123-00000008", "_NODEST=") in new stack
    -- Executing [0324771136@from-internal:5] Gosub("SIP/123123-00000008", "sub-record-check,s,1(out,0324771136,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/123123-00000008", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/123123-00000008", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("SIP/123123-00000008", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/123123-00000008", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("SIP/123123-00000008", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("SIP/123123-00000008", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("SIP/123123-00000008", "0?out,1") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/123123-00000008", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/123123-00000008", "NOW=1415873130") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/123123-00000008", "__DAY=13") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/123123-00000008", "__MONTH=11") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/123123-00000008", "__YEAR=2014") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/123123-00000008", "__TIMESTR=20141113-120530") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/123123-00000008", "__FROMEXTEN=123123") in new stack
    -- Executing [s@sub-record-check:21] Set("SIP/123123-00000008", "__CALLFILENAME=out-0324771136-123123-20141113-120530-1415873130.8") in new stack
    -- Executing [s@sub-record-check:22] Goto("SIP/123123-00000008", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/123123-00000008", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/123123-00000008", "0?record,1(exten,0324771136,123123)") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/123123-00000008", "") in new stack
    -- Executing [0324771136@from-internal:6] Macro("SIP/123123-00000008", "dialout-trunk,6,0324771136,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/123123-00000008", "DIAL_TRUNK=6") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/123123-00000008", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/123123-00000008", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/123123-00000008", "DIAL_NUMBER=0324771136") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/123123-00000008", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/123123-00000008", "OUTBOUND_GROUP=OUT_6") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/123123-00000008", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/123123-00000008", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/123123-00000008", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/123123-00000008", "0?sub-flp-6,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/123123-00000008", "OUTNUM=0324771136") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/123123-00000008", "custom=SIP/Ukrtelecom") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/123123-00000008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/123123-00000008", "0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/123123-00000008", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/123123-00000008", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/123123-00000008", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/123123-00000008", "1?Set(CONNECTEDLINE(num,i)=0324771136)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/123123-00000008", "1?Set(CONNECTEDLINE(name,i)=CID:123123)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/123123-00000008", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/123123-00000008", "SIP/Ukrtelecom/0324771136,300,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 18822
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0324771136@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 94.153.144.24:5060;branch=z9hG4bK230362e1;rport
Max-Forwards: 70
From: "123123" <sip:380892500614@sip.ukrtel.net>;tag=as11da731d
To: <sip:0324771136@sip.ukrtel.net>
Contact: <sip:380892500614@94.153.144.24:5060>
Call-ID: 182b59fe0a68f2be3f092ebf6b3832c6@sip.ukrtel.net
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.14.0)
Date: Thu, 13 Nov 2014 10:05:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 464179180 464179180 IN IP4 94.153.144.24
s=Asterisk PBX 11.14.0
c=IN IP4 94.153.144.24
t=0 0
m=audio 18822 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/Ukrtelecom/0324771136

<--- Transmitting (NAT) to 192.168.120.134:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;received=192.168.120.134;rport=5060
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as7704670a
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 INVITE
Server: FPBX-2.11.0(11.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0324771136@192.168.120.21:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 94.153.144.24:5060;received=94.153.144.27;branch=z9hG4bK230362e1;rport=5060
From: "123123" <sip:380892500614@sip.ukrtel.net>;tag=as11da731d
To: <sip:0324771136@sip.ukrtel.net>
Call-ID: 182b59fe0a68f2be3f092ebf6b3832c6@sip.ukrtel.net
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 94.153.144.24:5060;received=94.153.144.27;branch=z9hG4bK230362e1;rport=5060
From: "123123" <sip:380892500614@sip.ukrtel.net>;tag=as11da731d
To: <sip:0324771136@sip.ukrtel.net>;tag=aprqngfrt-86qn0630000c6
Call-ID: 182b59fe0a68f2be3f092ebf6b3832c6@sip.ukrtel.net
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0324771136@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 94.153.144.24:5060;branch=z9hG4bK230362e1;rport
Max-Forwards: 70
From: "123123" <sip:380892500614@sip.ukrtel.net>;tag=as11da731d
To: <sip:0324771136@sip.ukrtel.net>;tag=aprqngfrt-86qn0630000c6
Contact: <sip:380892500614@94.153.144.24:5060>
Call-ID: 182b59fe0a68f2be3f092ebf6b3832c6@sip.ukrtel.net
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.14.0)
Content-Length: 0


---
[2014-11-13 12:05:30] WARNING[31540][C-00000004]: chan_sip.c:23024 handle_response_invite: Received response: "Forbidden" from '"123123" <sip:380892500614@sip.ukrtel.net>;tag=as11da731d'
Scheduling destruction of SIP dialog '182b59fe0a68f2be3f092ebf6b3832c6@sip.ukrtel.net' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/123123-00000008", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/123123-00000008", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/123123-00000008", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/123123-00000008", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/123123-00000008", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/123123-00000008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] Set("SIP/123123-00000008", "CALLERID(number)=123123") in new stack
    -- Executing [0324771136@from-internal:7] Macro("SIP/123123-00000008", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/123123-00000008", "") in new stack
Audio is at 10886
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100011 (g726) to SDP

<--- Transmitting (NAT) to 192.168.120.134:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;received=192.168.120.134;rport=5060
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as7704670a
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 INVITE
Server: FPBX-2.11.0(11.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0324771136@192.168.120.21:5060>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1653871272 1653871272 IN IP4 192.168.120.21
s=Asterisk PBX 11.14.0
c=IN IP4 192.168.120.21
t=0 0
m=audio 10886 RTP/AVP 0 8 3 2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=ptime:20
a=sendrecv

<------------>
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/123123-00000008", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/123123-00000008", "1?intracompany,1") in new stack
    -- Goto (macro-outisbusy,intracompany,1)
    -- Executing [intracompany@macro-outisbusy:1] Playback("SIP/123123-00000008", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/123123-00000008 for all-circuits-busy-now&pls-try-call-later, noanswer
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory
[2014-11-13 12:05:30] WARNING[31674][C-00000004]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/123123-00000008 for all-circuits-busy-now&pls-try-call-later, noanswer
    -- Executing [intracompany@macro-outisbusy:2] Congestion("SIP/123123-00000008", "20") in new stack

<--- Reliably Transmitting (NAT) to 192.168.120.134:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;received=192.168.120.134;rport=5060
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as7704670a
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 INVITE
Server: FPBX-2.11.0(11.14.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
  == Spawn extension (macro-outisbusy, intracompany, 2) exited non-zero on 'SIP/123123-00000008' in macro 'outisbusy'
  == Spawn extension (from-internal, 0324771136, 7) exited non-zero on 'SIP/123123-00000008'
    -- Executing [h@from-internal:1] Hangup("SIP/123123-00000008", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/123123-00000008'

<--- SIP read from UDP:192.168.120.134:5060 --->
ACK sip:0324771136@192.168.120.21 SIP/2.0
Via: SIP/2.0/UDP 192.168.120.134:5060;branch=z9hG4bK001149448a69e411a83c373339fb1f74;rport
From: <sip:123123@192.168.120.21>;tag=3845589432
To: <sip:0324771136@192.168.120.21>;tag=as7704670a
Call-ID: 00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134
CSeq: 14 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '00114944-8A69-E411-A83A-373339FB1F74@192.168.120.134' Method: ACK
Reliably Transmitting (NAT) to 192.168.120.134:5060:
OPTIONS sip:123123@192.168.120.134:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.120.21:5060;branch=z9hG4bK0ecc3c62;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.120.21>;tag=as561f3410
To: <sip:123123@192.168.120.134:5060>
Contact: <sip:Unknown@192.168.120.21:5060>
Call-ID: 5f5cffdd45a57f5f4b954e0e4b3b7d50@192.168.120.21:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.14.0)
Date: Thu, 13 Nov 2014 10:05:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.120.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.120.21:5060;branch=z9hG4bK0ecc3c62;rport=5060
From: "Unknown" <sip:Unknown@192.168.120.21>;tag=as561f3410
To: <sip:123123@192.168.120.134:5060>;tag=80a7e1448a69e411a83c373339fb1f74
Call-ID: 5f5cffdd45a57f5f4b954e0e4b3b7d50@192.168.120.21:5060
CSeq: 102 OPTIONS
Contact: <sip:123123@192.168.120.134:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for phoner
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5f5cffdd45a57f5f4b954e0e4b3b7d50@192.168.120.21:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 195.5.0.83:5060:
OPTIONS sip:sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 94.153.144.24:5060;branch=z9hG4bK7e7a6d31;rport
Max-Forwards: 70
From: "Unknown" <sip:380892500614@94.153.144.24>;tag=as1b7c00b6
To: <sip:sip.ukrtel.net>
Contact: <sip:380892500614@94.153.144.24:5060>
Call-ID: 7c0154ce1359350a6c3f97205bc8bffa@94.153.144.24:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.14.0)
Date: Thu, 13 Nov 2014 10:05:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 94.153.144.24:5060;received=94.153.144.27;branch=z9hG4bK7e7a6d31;rport=5060
From: "Unknown" <sip:380892500614@sip.ukrtel.net>;tag=as1b7c00b6
To: <sip:sip.ukrtel.net>;tag=aprqngfrt-n4h62c20000c6
Call-ID: 7c0154ce1359350a6c3f97205bc8bffa@94.153.144.24:5060
CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '7c0154ce1359350a6c3f97205bc8bffa@94.153.144.24:5060' Method: OPTIONS

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1. "Asterisk 11 - регистрация транка"  +/
Сообщение от suslic (ok) on 13-Ноя-14, 16:45 
>[оверквотинг удален]
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 94.153.144.24:5060;received=94.153.144.27;branch=z9hG4bK7e7a6d31;rport=5060
> From: "Unknown" <sip:380892500614@sip.ukrtel.net>;tag=as1b7c00b6
> To: <sip:sip.ukrtel.net>;tag=aprqngfrt-n4h62c20000c6
> Call-ID: 7c0154ce1359350a6c3f97205bc8bffa@94.153.144.24:5060
> CSeq: 102 OPTIONS
> <------------->
> --- (6 headers 0 lines) ---
> Really destroying SIP dialog '7c0154ce1359350a6c3f97205bc8bffa@94.153.144.24:5060'
> Method: OPTIONS

Заработало с регистрацией
register => user:password@sip.ukrtel.net

Баз регистрации не запустилось - провайдер Укртелеком

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