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Не прохдят исходящие звонки Asterisk (Ver. 11.16.0) + freepbx

Newbies/FAQ Forum 7 сообщений -
#1
Здравствуйте.
Имеется Asterisk (Ver. 11.16.0) + freepbx установленный из дистрибутива AsteriskNow
Имеется ip телефония от мегафона, оператором была предоставлена для подключения следующая информация
73432375050/ XXXXX (где XXXXX это пароль)
sip server: 217.115.80.105;
SIP-domen: ekb.synterra-ural.ru

В Asterisk настроен транк

PEER Details:
disallow=all
type=peer
username=73432375050
secret=XXXXX
host=217.115.80.105
insecure=port,invite
port=5060
fromuser=73432375050
context=from-trunk
dtmfmode=inband
allow=alaw

Register String
73432375050@ekb.synterra-ural.ru:XXXXX :73432375050@217.115.80.105/73432375050

Так же был подключен IP телефон Yealink с внутренним номером 101


После всех этих манипуляций входящие звонки на телефон проходят, а вот исходящие не проходят, при попытке позвонить на любой номер в трубке слышно "The number you have dialed is not in service. Please check the number and try again." (Номер, который вы набрали, в настоящий момент не работает. Пожалуйста, проверьте номер и попробуйте еще раз.)

Лог во время исходящего звонка на номер 89221221481

Connected to Asterisk 11.16.0 currently running on localhost (pid = 2032)







INVITE sip:89221221481@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK4212574985
From: "Мегафон" ;tag=1867317660
To:
Call-ID: 1246034179@192.168.2.201
CSeq: 1 INVITE
Contact:
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T19P 31.72.14.5
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 308

v=0
o=- 20062 20062 IN IP4 192.168.2.201
s=SDP data
c=IN IP4 192.168.2.201
t=0 0
m=audio 11794 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

--- (14 headers 15 lines) ---
Sending to 192.168.2.201:5062 (NAT)
Sending to 192.168.2.201:5062 (NAT)
Using INVITE request as basis request - 1246034179@192.168.2.201
Found peer '101' for '101' from 192.168.2.201:5062


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK4212574985;received=192.168.2.201;rport=5062
From: "Мегафон" ;tag=1867317660
To: ;tag=as49641657
Call-ID: 1246034179@192.168.2.201
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b2a6cfc"
Content-Length: 0



Scheduling destruction of SIP dialog '1246034179@192.168.2.201' in 6464 ms (Method: INVITE)


ACK sip:89221221481@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK4212574985
From: "Мегафон" ;tag=1867317660
To: ;tag=as49641657
Call-ID: 1246034179@192.168.2.201
CSeq: 1 ACK
Content-Length: 0


--- (7 headers 0 lines) ---


INVITE sip:89221221481@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163
From: "Мегафон" ;tag=1867317660
To:
Call-ID: 1246034179@192.168.2.201
CSeq: 2 INVITE
Contact:
Authorization: Digest username="101", realm="asterisk", nonce="1b2a6cfc", uri="sip:89221221481@192.168.2.3", response="144a5ab3b9587634f2fd081319a4ea5b", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T19P 31.72.14.5
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 308

v=0
o=- 20062 20062 IN IP4 192.168.2.201
s=SDP data
c=IN IP4 192.168.2.201
t=0 0
m=audio 11794 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

--- (15 headers 15 lines) ---
Sending to 192.168.2.201:5062 (NAT)
Using INVITE request as basis request - 1246034179@192.168.2.201
Found peer '101' for '101' from 192.168.2.201:5062
== 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 18
Found RTP audio format 9
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 G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726|g729), peer - audio=(ulaw|alaw|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.201:11794
Looking for 89221221481 in from-internal (domain 192.168.2.3)
list_route: hop:


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163;received=192.168.2.201;rport=5062
From: "Мегафон" ;tag=1867317660
To:
Call-ID: 1246034179@192.168.2.201
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
Content-Length: 0



-- Executing [89221221481@from-internal:1] Macro("SIP/101-00000345", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/101-00000345", "TOUCH_MONITOR=1446319622.837") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/101-00000345", "AMPUSER=101") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-00000345", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/101-00000345", "1?Set(REALCALLERIDNUM=101)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/101-00000345", "AMPUSER=101") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000345", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/101-00000345", "AMPUSERCIDNAME=101") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/101-00000345", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/101-00000345", "AMPUSERCID=101") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/101-00000345", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/101-00000345", "CALLERID(all)="101" ") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/101-00000345", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/101-00000345", "1?Set(GROUP(concurrency_limit)=101)") in new stack
-- Executing [s@macro-user-callerid:14] GosubIf("SIP/101-00000345", "7?sub-ccss,s,1(from-internal,89221221481)") in new stack
-- Executing [s@sub-ccss:1] ExecIf("SIP/101-00000345", "0?Return()") in new stack
-- Executing [s@sub-ccss:2] Set("SIP/101-00000345", "CCSS_SETUP=TRUE") in new stack
-- Executing [s@sub-ccss:3] GosubIf("SIP/101-00000345", "0?monitor_config,1(from-internal,89221221481):monitor_default,1(from-internal,89221221481)") in new stack
-- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/101-00000345", "0?is_exten") in new stack
-- Executing [monitor_default@sub-ccss:2] StackPop("SIP/101-00000345", "") in new stack
-- Executing [monitor_default@sub-ccss:3] Return("SIP/101-00000345", "FALSE") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("SIP/101-00000345", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/101-00000345", "1?continue") in new stack
-- Goto (macro-user-callerid,s,30)
-- Executing [s@macro-user-callerid:30] Set("SIP/101-00000345", "CALLERID(number)=101") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/101-00000345", "CALLERID(name)=101") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/101-00000345", "CDR(cnum)=101") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/101-00000345", "CDR(cnam)=101") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/101-00000345", "CHANNEL(language)=en") in new stack
-- Executing [89221221481@from-internal:2] Gosub("SIP/101-00000345", "sub-record-check,s,1(out,89221221481,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/101-00000345", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/101-00000345", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/101-00000345", "NOW=1446319622") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/101-00000345", "__DAY=31") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/101-00000345", "__MONTH=10") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/101-00000345", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/101-00000345", "__TIMESTR=20151031-142702") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/101-00000345", "__FROMEXTEN=101") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/101-00000345", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/101-00000345", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/101-00000345", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/101-00000345", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/101-00000345", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/101-00000345", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/101-00000345", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/101-00000345", "Outbound Recording Check from 101 to 89221221481") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/101-00000345", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/101-00000345", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/101-00000345", "recordcheck,1(dontcare,out,89221221481)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/101-00000345", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/101-00000345", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/101-00000345", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/101-00000345", "") in new stack
-- Executing [89221221481@from-internal:3] ExecIf("SIP/101-00000345", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [89221221481@from-internal:4] Set("SIP/101-00000345", "MOHCLASS=default") in new stack
-- Executing [89221221481@from-internal:5] ExecIf("SIP/101-00000345", "1?Set(TRUNKCIDOVERRIDE=73432375050)") in new stack
-- Executing [89221221481@from-internal:6] Set("SIP/101-00000345", "_NODEST=") in new stack
-- Executing [89221221481@from-internal:7] Macro("SIP/101-00000345", "dialout-trunk,2,89221221481,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000345", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000345", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000345", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000345", "DIAL_NUMBER=89221221481") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000345", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000345", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000345", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000345", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000345", "DIAL_TRUNK_OPTIONS=Tt") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000345", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000345", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000345", "0?Set(REALCALLERIDNUM=101)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000345", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000345", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000345", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000345", "TRUNKOUTCID=73432375050") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000345", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000345", "1?Set(CALLERID(all)=73432375050)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000345", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/101-00000345", "1?Set(CALLERID(all)=73432375050)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/101-00000345", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/101-00000345", "CDR(outbound_cnum)=73432375050") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/101-00000345", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000345", "0?sub-flp-2,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000345", "OUTNUM=89221221481") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000345", "custom=SIP/mvoip1") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000345", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/101-00000345", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/101-00000345", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000345", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000345", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/101-00000345", "1?Set(CONNECTEDLINE(num,i)=89221221481)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/101-00000345", "1?Set(CONNECTEDLINE(name,i)=CID:73432375050)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/101-00000345", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/101-00000345", "SIP/mvoip1/89221221481,300,Tt") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 15492
Adding codec 100004 (alaw) to SDP
Reliably Transmitting (NAT) to 217.115.80.105:5060:
INVITE sip:89221221481@217.115.80.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK6b254f69;rport
Max-Forwards: 70
From: ;tag=as0365b4c6
To:
Contact:
Call-ID: 014ef6280f58e09d0448af59534d22a7@192.168.2.3:5060
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Date: Sat, 31 Oct 2015 19:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 178

v=0
o=root 1035201604 1035201604 IN IP4 192.168.2.3
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.2.3
t=0 0
m=audio 15492 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
-- Called SIP/mvoip1/89221221481


SIP/2.0 100 Trying
From: ;tag=as0365b4c6
To:
Call-ID: 014ef6280f58e09d0448af59534d22a7@192.168.2.3:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK6b254f69;rport=5060;received=188.234.216.98
Content-Length: 0


--- (7 headers 0 lines) ---


SIP/2.0 404 Not Found
From: ;tag=as0365b4c6
To: ;tag=1940650a-13c4-5634892d-a98fa816-2df9c549
Call-ID: 014ef6280f58e09d0448af59534d22a7@192.168.2.3:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK6b254f69;rport=5060;received=188.234.216.98
Content-Length: 0


--- (7 headers 0 lines) ---
Transmitting (NAT) to 217.115.80.105:5060:
ACK sip:89221221481@217.115.80.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK6b254f69;rport
Max-Forwards: 70
From: ;tag=as0365b4c6
To: ;tag=1940650a-13c4-5634892d-a98fa816-2df9c549
Contact:
Call-ID: 014ef6280f58e09d0448af59534d22a7@192.168.2.3:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '014ef6280f58e09d0448af59534d22a7@192.168.2.3:5060' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/101-00000345", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/101-00000345", "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/101-00000345", "RC=1") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000345", "1,1") in new stack
-- Goto (macro-dialout-trunk,1,1)
-- Executing [1@macro-dialout-trunk:1] Goto("SIP/101-00000345", "s-INVALIDNMBR,1") in new stack
-- Goto (macro-dialout-trunk,s-INVALIDNMBR,1)
-- Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp("SIP/101-00000345", "Dial failed due to trunk reporting Address Incomplete - giving up") in new stack
-- Executing [s-INVALIDNMBR@macro-dialout-trunk:2] Progress("SIP/101-00000345", "") in new stack
Audio is at 13684
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163;received=192.168.2.201;rport=5062
From: "Мегафон" ;tag=1867317660
To: ;tag=as5b1a1a7c
Call-ID: 1246034179@192.168.2.201
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 1831680709 1831680709 IN IP4 192.168.2.3
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.2.3
t=0 0
m=audio 13684 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Executing [s-INVALIDNMBR@macro-dialout-trunk:3] Playback("SIP/101-00000345", "ss-noservice,noanswer") in new stack
-- Playing 'ss-noservice.ulaw' (language 'en')
> 0xb58c4f78 -- Probation passed - setting RTP source address to 192.168.2.201:11794


CANCEL sip:89221221481@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163
From: "Мегафон" ;tag=1867317660
To:
Call-ID: 1246034179@192.168.2.201
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Yealink SIP-T19P 31.72.14.5
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 192.168.2.201:5062 (NAT)


SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163;received=192.168.2.201;rport=5062
From: "Мегафон" ;tag=1867317660
To: ;tag=as5b1a1a7c
Call-ID: 1246034179@192.168.2.201
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0





SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163;received=192.168.2.201;rport=5062
From: "Мегафон" ;tag=1867317660
To: ;tag=as5b1a1a7c
Call-ID: 1246034179@192.168.2.201
CSeq: 2 CANCEL
Server: FPBX-AsteriskNOW-12.0.76.2(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0



== Spawn extension (macro-dialout-trunk, s-INVALIDNMBR, 3) exited non-zero on 'SIP/101-00000345' in macro 'dialout-trunk'
== Spawn extension (from-internal, 89221221481, 7) exited non-zero on 'SIP/101-00000345'
-- Executing [h@from-internal:1] Hangup("SIP/101-00000345", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000345'


ACK sip:89221221481@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5062;branch=z9hG4bK1084137163
From: "Мегафон" ;tag=1867317660
To: ;tag=as5b1a1a7c
Call-ID: 1246034179@192.168.2.201
CSeq: 2 ACK
Content-Length: 0


Подскажите пожалуйста, в какую сторону копать, несколько дней не могу решить эту проблему. Спасибо.
#2
От вашего оператора вам приходит ответ:

Цитата:

SIP/2.0 404 Not Found


что скорее всего означает, что вы набираете номер не в том формате. Обратитесь за деталями к вашему оператору.
#3
Asterisk за NAT?
Чего в обмене все время локальный IP адрес? настройте правильно.
#4
Пообщался с техподдержкой мегафона, получил ответ "В сообщениях INVITE, в приходящих от вас вызовах не указывается домен ekb.synterra-ural.ru ."

что происходит при вызове сейчас:

[2015-11-04 04:23:01] VERBOSE[4434][C-00000005] app_dial.c: -- Called SIP/mvoip1/89221221481
[2015-11-04 04:23:01] VERBOSE[2035] chan_sip.c:

SIP/2.0 100 Trying
From: ;tag=as6b418673
To:
Call-ID: 68468952259f2f6770d734e530c62d64@ekb.synterra-ural.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 188.234.251.116:5060;branch=z9hG4bK4a12e05b;rport=5060
Content-Length: 0


[2015-11-04 04:23:01] VERBOSE[2035] chan_sip.c: --- (7 headers 0 lines) ---
[2015-11-04 04:23:01] VERBOSE[2035] chan_sip.c:

SIP/2.0 404 Not Found
From: ;tag=as6b418673
To: ;tag=1940650a-13c4-5639ce75-be26231c-21876bf1
Call-ID: 68468952259f2f6770d734e530c62d64@ekb.synterra-ural.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 188.234.251.116:5060;branch=z9hG4bK4a12e05b;rport=5060
Content-Length: 0

Я правильно понимаю, то что выделено жирным нужно заменить на ekb.synterra-ural.ru?
Как это сделать с помощью веб интерфейса FreePBX? или же нужно править extensions.conf?
спасибо за помощь.
#5
Вы показали ответы, а нужно было показать запросы.
В вашем случае должно помочь
Код:
fromdomain=ekb.synterra-ural.ru

в настроках пира
#6
Цитата:
insecure=port,invite

Сомневаюсь что не insecure=invite.
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#7
Проблема решена, исходящие заработали.
Решение проблемы заключалось вот в следующем:
В поле host, в настройках PEER Details нужно вписать host=sbc.synterra-ural.ru, причем не ekb.synterra-ural.ru, а именно sbc.synterra-ural.ru, ну или ekb.synterra-ural.ru добавить в настройки днс, потому что так оно у меня не резолвилось.
Всем спасибо за помощь.