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Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

https://forum.asterisk.ru/viewtopic.php?t=12837

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Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 15:12
Dimenko
Добрый день. Настраиваем Freepbx - sip транк на провайдера Dom.ru - регистрация проходит, входящие звонки принимает, исходящие - никак. Через софтфон всё ок.


Логи:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:192.168.10.209:65325 --->
INVITE sip:+74997554129@10.2.90.43:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---c4dc9479e7977709;rport
Max-Forwards: 70
Contact: <sip:4001@192.168.10.209:65325;rinstance=4d0e1e9fba01d7de>
To: <sip:+74997554129@10.2.90.43:5062>
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 1 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 338

v=0
o=- 13188481669296749 1 IN IP4 192.168.10.209
s=X-Lite release 5.4.0 stamp 94388
c=IN IP4 192.168.10.209
t=0 0
m=audio 54708 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 192.168.10.209:65325 (NAT)
Sending to 192.168.10.209:65325 (NAT)
Using INVITE request as basis request - 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
Found peer '4001' for '4001' from 192.168.10.209:65325

<--- Reliably Transmitting (no NAT) to 192.168.10.209:65325 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---c4dc9479e7977709;received=192.168.10.209;rport=65325
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
To: <sip:+74997554129@10.2.90.43:5062>;tag=as111a8978
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 1 INVITE
Server: FPBX-14.0.5.2(15.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="056784cb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.10.209:65325 --->
ACK sip:+74997554129@10.2.90.43:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---c4dc9479e7977709;rport
Max-Forwards: 70
To: <sip:+74997554129@10.2.90.43:5062>;tag=as111a8978
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.10.209:65325 --->
INVITE sip:+74997554129@10.2.90.43:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---b4b26078daa14e52;rport
Max-Forwards: 70
Contact: <sip:4001@192.168.10.209:65325;rinstance=4d0e1e9fba01d7de>
To: <sip:+74997554129@10.2.90.43:5062>
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 2 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Authorization: Digest username="4001",realm="asterisk",nonce="056784cb",uri="sip:+74997554129@10.2.90.43:5062",response="49266316980fea947fa78cf222ed607d",algorithm=MD5
Content-Length: 338

v=0
o=- 13188481669296749 1 IN IP4 192.168.10.209
s=X-Lite release 5.4.0 stamp 94388
c=IN IP4 192.168.10.209
t=0 0
m=audio 54708 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.10.209:65325 (no NAT)
Using INVITE request as basis request - 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
Found peer '4001' for '4001' from 192.168.10.209:65325
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 120
Found RTP audio format 0
Found RTP audio format 84
Found RTP audio format 101
Found audio description format opus for ID 120
Found audio description format speex for ID 84
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722|g719), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f963401d980 -- Strict RTP learning after remote address set to: 192.168.10.209:54708
Peer audio RTP is at port 192.168.10.209:54708
Looking for +74997554129 in from-internal (domain 10.2.90.43)
sip_route_dump: route/path hop: <sip:4001@192.168.10.209:65325;rinstance=4d0e1e9fba01d7de>

<--- Transmitting (no NAT) to 192.168.10.209:65325 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---b4b26078daa14e52;received=192.168.10.209;rport=65325
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
To: <sip:+74997554129@10.2.90.43:5062>
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 2 INVITE
Server: FPBX-14.0.5.2(15.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+74997554129@10.2.90.43:5062>
Content-Length: 0


<------------>
-- Executing [+74997554129@from-internal:1] Macro("SIP/4001-00000005", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/4001-00000005", "TOUCH_MONITOR=1544008069.9") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/4001-00000005", "AMPUSER=4001") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/4001-00000005", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/4001-00000005", "1?Set(REALCALLERIDNUM=4001)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/4001-00000005", "AMPUSER=4001") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/4001-00000005", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/4001-00000005", "AMPUSERCIDNAME=4001") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("SIP/4001-00000005", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/4001-00000005", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/4001-00000005", "AMPUSERCID=4001") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/4001-00000005", "__DIAL_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/4001-00000005", "CALLERID(all)="4001" <4001>") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/4001-00000005", "0?limit") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/4001-00000005", "1?Set(GROUP(concurrency_limit)=4001)") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("SIP/4001-00000005", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:16] NoOp("SIP/4001-00000005", "Macro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/4001-00000005", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s@macro-user-callerid:18] GotoIf("SIP/4001-00000005", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [s@macro-user-callerid:37] Set("SIP/4001-00000005", "CALLERID(number)=4001") in new stack
-- Executing [s@macro-user-callerid:38] Set("SIP/4001-00000005", "CALLERID(name)=4001") in new stack
-- Executing [s@macro-user-callerid:39] GotoIf("SIP/4001-00000005", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:40] Set("SIP/4001-00000005", "CDR(cnam)=4001") in new stack
-- Executing [s@macro-user-callerid:41] Set("SIP/4001-00000005", "CDR(cnum)=4001") in new stack
-- Executing [s@macro-user-callerid:42] Set("SIP/4001-00000005", "CHANNEL(language)=en") in new stack
-- Executing [+74997554129@from-internal:2] Gosub("SIP/4001-00000005", "sub-record-check,s,1(out,+74997554129,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/4001-00000005", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/4001-00000005", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/4001-00000005", "NOW=1544008069") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/4001-00000005", "__DAY=05") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/4001-00000005", "__MONTH=12") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/4001-00000005", "__YEAR=2018") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/4001-00000005", "__TIMESTR=20181205-140749") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/4001-00000005", "__FROMEXTEN=4001") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/4001-00000005", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/4001-00000005", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/4001-00000005", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/4001-00000005", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/4001-00000005", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/4001-00000005", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/4001-00000005", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/4001-00000005", "Outbound Recording Check from 4001 to +74997554129") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/4001-00000005", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/4001-00000005", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/4001-00000005", "recordcheck,1(dontcare,out,+74997554129)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/4001-00000005", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/4001-00000005", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/4001-00000005", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/4001-00000005", "") in new stack
-- Executing [+74997554129@from-internal:3] ExecIf("SIP/4001-00000005", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [+74997554129@from-internal:4] Set("SIP/4001-00000005", "MOHCLASS=default") in new stack
-- Executing [+74997554129@from-internal:5] Set("SIP/4001-00000005", "_NODEST=") in new stack
-- Executing [+74997554129@from-internal:6] Macro("SIP/4001-00000005", "dialout-trunk,7,+74997554129,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/4001-00000005", "DIAL_TRUNK=7") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/4001-00000005", "0?Set(DIAL_OPTIONS=Hhtr)") in new stack
-- Executing [s@macro-dialout-trunk:3] GosubIf("SIP/4001-00000005", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:4] ExecIf("SIP/4001-00000005", "0?Set(CALLERID(num)=4001)") in new stack
-- Executing [s@macro-dialout-trunk:5] GotoIf("SIP/4001-00000005", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/4001-00000005", "DIAL_NUMBER=+74997554129") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("SIP/4001-00000005", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("SIP/4001-00000005", "OUTBOUND_GROUP=OUT_7") in new stack
-- Executing [s@macro-dialout-trunk:9] Set("SIP/4001-00000005", "DIAL_TRUNK_OPTIONS=T") in new stack
-- Executing [s@macro-dialout-trunk:10] GotoIf("SIP/4001-00000005", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,12)
-- Executing [s@macro-dialout-trunk:12] GotoIf("SIP/4001-00000005", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:13] Macro("SIP/4001-00000005", "outbound-callerid,7") in new stack
-- Executing [s@macro-outbound-callerid:1] NoOp("SIP/4001-00000005", "4001") in new stack
-- Executing [s@macro-outbound-callerid:2] NoOp("SIP/4001-00000005", "") in new stack
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/4001-00000005", "off") in new stack
-- Executing [s@macro-outbound-callerid:4] ExecIf("SIP/4001-00000005", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:5] ExecIf("SIP/4001-00000005", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:6] ExecIf("SIP/4001-00000005", "0?Set(REALCALLERIDNUM=4001)") in new stack
-- Executing [s@macro-outbound-callerid:7] ExecIf("SIP/4001-00000005", "0?Set(AMPUSER=4001)") in new stack
-- Executing [s@macro-outbound-callerid:8] GotoIf("SIP/4001-00000005", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] Set("SIP/4001-00000005", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:13] Set("SIP/4001-00000005", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:14] Set("SIP/4001-00000005", "TRUNKOUTCID=74742565235") in new stack
-- Executing [s@macro-outbound-callerid:15] GotoIf("SIP/4001-00000005", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,20)
-- Executing [s@macro-outbound-callerid:20] ExecIf("SIP/4001-00000005", "1?Set(CALLERID(all)=74742565235)") in new stack
-- Executing [s@macro-outbound-callerid:21] ExecIf("SIP/4001-00000005", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:22] ExecIf("SIP/4001-00000005", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:23] ExecIf("SIP/4001-00000005", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:24] ExecIf("SIP/4001-00000005", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:25] Set("SIP/4001-00000005", "CDR(outbound_cnum)=74742565235") in new stack
-- Executing [s@macro-outbound-callerid:26] Set("SIP/4001-00000005", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:14] GosubIf("SIP/4001-00000005", "0?sub-flp-7,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:15] Set("SIP/4001-00000005", "OUTNUM=+74997554129") in new stack
-- Executing [s@macro-dialout-trunk:16] Set("SIP/4001-00000005", "custom=PJSIP") in new stack
-- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/4001-00000005", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
-- Executing [s@macro-dialout-trunk:18] ExecIf("SIP/4001-00000005", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:19] Macro("SIP/4001-00000005", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4001-00000005", "") in new stack
-- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/4001-00000005", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("SIP/4001-00000005", "__CRM_DIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("SIP/4001-00000005", "__CRM_DESTINATION=+74997554129") in new stack
-- Executing [s@macro-dialout-trunk:23] Set("SIP/4001-00000005", "__CRM_SOURCE=4001") in new stack
-- Executing [s@macro-dialout-trunk:24] AGI("SIP/4001-00000005", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/4001-00000005>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:25] Set("SIP/4001-00000005", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:26] NoOp("SIP/4001-00000005", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:27] GotoIf("SIP/4001-00000005", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/4001-00000005", "1?Set(CONNECTEDLINE(num,i)=+74997554129)") in new stack
-- Executing [s@macro-dialout-trunk:29] ExecIf("SIP/4001-00000005", "1?Set(CONNECTEDLINE(name,i)=CID:74742565235)") in new stack
-- Executing [s@macro-dialout-trunk:30] ExecIf("SIP/4001-00000005", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)74742565235)") in new stack
-- Executing [s@macro-dialout-trunk:31] GotoIf("SIP/4001-00000005", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:32] Dial("SIP/4001-00000005", "PJSIP/+74997554129@test_short3,300,Tb(func-apply-sipheaders^s^1)") in new stack
-- PJSIP/test_short3-00000004 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/test_short3-00000004", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/test_short3-00000004", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("PJSIP/test_short3-00000004", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:4] ExecIf("PJSIP/test_short3-00000004", "0?Set(Rheader=1)") in new stack
-- Executing [s@func-apply-sipheaders:5] While("PJSIP/test_short3-00000004", "0") in new stack
-- Jumping to priority 9
-- Executing [s@func-apply-sipheaders:10] ExecIf("PJSIP/test_short3-00000004", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [s@func-apply-sipheaders:11] ExecIf("PJSIP/test_short3-00000004", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [s@func-apply-sipheaders:12] Return("PJSIP/test_short3-00000004", "") in new stack
== Spawn extension (from-pstn, +74997554129, 1) exited non-zero on 'PJSIP/test_short3-00000004'
-- PJSIP/test_short3-00000004 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called PJSIP/+74997554129@test_short3
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:33] NoOp("SIP/4001-00000005", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:34] GotoIf("SIP/4001-00000005", "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/4001-00000005", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/4001-00000005", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/4001-00000005", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/4001-00000005", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/4001-00000005", "1?Set(CALLERID(number)=4001)") in new stack
-- Executing [+74997554129@from-internal:7] Macro("SIP/4001-00000005", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/4001-00000005", "") in new stack
Audio is at 10458
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.10.209:65325 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---b4b26078daa14e52;received=192.168.10.209;rport=65325
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
To: <sip:+74997554129@10.2.90.43:5062>;tag=as4cf800fd
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 2 INVITE
Server: FPBX-14.0.5.2(15.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+74997554129@10.2.90.43:5062>
Content-Type: application/sdp
Content-Length: 295

v=0
o=root 1477910874 1477910874 IN IP4 10.2.90.43
s=Asterisk PBX 15.5.0
c=IN IP4 10.2.90.43
t=0 0
m=audio 10458 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/4001-00000005", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/4001-00000005", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/4001-00000005", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
-- <SIP/4001-00000005> Playing 'all-circuits-busy-now.ulaw' (language 'en')
> 0x7f963401d980 -- Strict RTP switching to RTP target address 192.168.10.209:54708 as source
-- <SIP/4001-00000005> Playing 'please-try-call-later.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/4001-00000005", "20") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.10.209:65325 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---b4b26078daa14e52;received=192.168.10.209;rport=65325
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
To: <sip:+74997554129@10.2.90.43:5062>;tag=as4cf800fd
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 2 INVITE
Server: FPBX-14.0.5.2(15.5.0)
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


<------------>
[2018-12-05 14:07:53] WARNING[125563][C-00000006]: channel.c:4875 ast_prod: Prodding channel 'SIP/4001-00000005' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/4001-00000005' in macro 'outisbusy'
== Spawn extension (from-internal, +74997554129, 7) exited non-zero on 'SIP/4001-00000005'
-- Executing [h@from-internal:1] Macro("SIP/4001-00000005", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/4001-00000005", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/4001-00000005", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("SIP/4001-00000005", " monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("SIP/4001-00000005", "attendedtransfer-rec-restart.php,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <SIP/4001-00000005>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("SIP/4001-00000005", "") in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/4001-00000005' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4001-00000005'
-- SIP/4001-00000005 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/4001-00000005", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/4001-00000005", "HANGUP CAUSE: 34") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/4001-00000005", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/4001-00000005", "MASTER CHANNEL: 1544008069.9 = 1544008069.9") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/4001-00000005", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/4001-00000005", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/4001-00000005", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi

<--- SIP read from UDP:192.168.10.209:65325 --->
ACK sip:+74997554129@10.2.90.43:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.209:65325;branch=z9hG4bK-524287-1---b4b26078daa14e52;rport
Max-Forwards: 70
To: <sip:+74997554129@10.2.90.43:5062>;tag=as4cf800fd
From: "Desk"<sip:4001@10.2.90.43:5062>;tag=34b6bf60
Call-ID: 94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
-- <SIP/4001-00000005>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/4001-00000005", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4001-00000005'
-- SIP/4001-00000005 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
Really destroying SIP dialog '94388YTM1MzI0ODQ3YTA2NzViODI5ZGNjMGUwMGRiMDJiMDY' Method: ACK

<--- SIP read from UDP:192.168.10.209:65325 --->
pjsip.endpoint.conf
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
#include pjsip.endpoint_custom.conf

[anonymous]
type=endpoint
context=from-sip-external
allow=all

[test_short3]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=alaw,ulaw,gsm,g726,g722,g719,g723,speex,speex16,speex32,siren7,adpcm,silk8,silk12,silk16,silk24
aors=test_short3
language=en
outbound_auth=test_short3
from_domain=voip.domru.ru
from_user=74742565235
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
direct_media=no
rewrite_contact=yes
rtp_symmetric=yes
dtmf_mode=auto

[dpma_endpoint]
type=endpoint
context=dpma-invalid

Re: Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 16:13
ded
Настройте через chan_sip вместо pjsip и будет вам профит.
Вам же ездить, а не шашечки?

Re: Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 17:10
chardisdze
У dom.ru фишка есть в Нижегородской области точно. Строка регистрации видоизмененная. Решил запросом строки у ТП.
Хотя это во входящих

Re: Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 17:18
Dimenko
с радостью бы, но chan_sip вообще ни в какую...регистрация не проходит даже
собственно, входные данные такие:
Номер: 565235
User name: 74742565235
Authorization User Name: 565235
Password: xxxx

Общие рекомендации от провайдера:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Общие рекомендации:
1) Поле User Name заполняется в формате е164 (7<код города><номер абонента>).
2) Обязательно использование поля Authorization User ID (Authorization User Name, Auth ID) в формате subscriber (<номер абонента>).
3) Должен быть разрешен прием сигнальных сообщений SIP с voip.domru.ru
4) Отправка сигнальных сообщений SIP должна осуществляться на voip.domru.ru
5) Оборудование не должно быть расположено за NAT. В подобном случае работоспособность услуги не гарантируется.
6) Использование кодеков G711A-law, G.729.
7) Использование протокола SIP v.2.0 (RFC 2543bis\3261)
8) Использование транспортного протокола UDP
9) Передача факсов по протоколу T.38
10) Трансляция DTMF по стандарту RFC2833
11) Формат номера A: E164 (7<код города><номер абонента>).
12) Формат номера B: E164 . Вызовы, начинающиеся с 7 – отправляются на оператора IP-телефонии. Вызовы, начинающиеся с 8 – отправляются через ТФоП.
Комбинации конфигов пробовали разные, примерно такие:

type=peer
secret=xxxx
username=565235
fromuser=74742565235
fromdomain=188.234.136.49
host=188.234.136.49
realm=188.234.136.49
disallow=all
allow=alaw,ulaw
insecure=invite
callerid=565235
dtmfmode=rfc2833
-----------------------
username=565235
type=peer
secret=xxxx
nat=no
insecure=invite
host=voip.domru.ru
fromuser=74742565235
fromdomain=voip.domru.ru:5060
disallow=all
allow=alaw

Строки регистрации не пропускают в любых комбинациях с username 565235
[2018-12-05 16:57:42] WARNING[5367]: chan_sip.c:24544 handle_response_register: Got 404 Not found on SIP register to service 565235@voip.domru.ru, giving up

Провайдер утверждает, что строка регистрации должна выглядеть следующим образом:
74742565235:565235:xxxx@voip.domru.ru:5060/74742565235

Но при такой строке и с другими вариациями с username 74742565235 выбивает timeout и Auth. sent в Статусе.

Было бы супер, если бы подсказали, как можно адаптировать частично работающие параметры из pjsip для sip.

Re: Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 18:43
zzuz
"Got 404 Not found on SIP register to service"
Проверьте , что адрес voip.domru.ru корректно резолвится.

И строка должна быть
74742565235:xxxxxx:565235@voip.domru.ru:5060/74742565235

Re: Не работают исходящие вызовы (FreePBX, провайдер Dom.ru)

Добавлено: 05 дек 2018, 18:49
zzuz
По рекомендациям от провайдера пир должен быть описан , как

Код: Выделить всё

type=peer
secret=xxxx
username=74742565235
fromuser=565235
fromdomain=voip.domru.ru
host=voip.domru.ru
disallow=all
allow=alaw,ulaw
insecure=invite
callerid=565235
dtmfmode=rfc2833
Поле USER не нужно , его можно убрать, а то у вас два type=peer с одними и теми же параметрами.
Часовой пояс: UTC+03:00
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