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FreePBX и исходящие

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
sergeev2d
Сообщения: 3
Зарегистрирован: 24 ноя 2014, 11:04

FreePBX и исходящие

Сообщение sergeev2d »

Здравствуйте.
Знаю вопрос извечный и постоянный. Прочитал весь форум и ничего не помогает.
Данные по теме:
настроено так - SIP провайдер <-> Net <-> шлюз + Asterisk <-> Asterisk настраиваемый.
Порты проброшены.
На вопрос зачем так, кроме как надо ничего сказать не могу)
Настройки пира даны провайдером:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
type=friend
host=79.*.*.*
username=*******
fromuser=******
quality=1000
callerid=**********
bindaddr=79.*.*.*
tcpbindaddr=79.*.*.*
sip_general_custom
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
allowguest=no
allowoverlap=no
bindport=5060
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=no
dtmfmode=inband
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
externip=213.*.*.*
localnet=192.168.0.0/24
nat=route
Через данного провайдера нет звука при исходящем звонке. Меня смущает что при звонке происходит следующее:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000525, ts 3819642001, len 000160)
Sent RTP packet to 10.*.*.*:18206 (type 08, seq 006056, ts 3819642000, len 000160)
-- Started music on hold, class 'default', on SIP/200-00000000
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000526, ts 3819642161, len 000160)
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000527, ts 3819642321, len 000160)
Sent RTP packet to 192.168.0.77:20824 (type 08, seq 027574, ts 000160, len 000160)
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000606, ts 3819654961, len 000160)
-- Stopped music on hold on SIP/200-00000000
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000607, ts 3819655121, len 000160)
Sent RTP packet to 79.*.*.*:16484 (type 08, seq 006057, ts 3819655120, len 000160)
Got RTP packet from 192.168.0.77:20824 (type 08, seq 000608, ts 3819655281, len 000160)
10.*.*. - это провайдерский прокси.
На всякий случай debug peer:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
localhost*CLI> sip set debug peer 200
SIP Debugging Enabled for IP: 192.168.0.77

<--- SIP read from UDP:192.168.0.77:6855 --->
INVITE sip:************@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;rport
Max-Forwards: 70
Contact: <sip:200@192.168.0.77:6855>;+sip.instance="<urn:uuid:************>"
To: <sip:************@192.168.0.68>
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: PortGo v8.1, Build 01220214
Content-Length: 241

v=0
o=- 8314748 8314748 IN IP4 192.168.0.77
s=http://www.portsip.com
c=IN IP4 192.168.0.77
t=0 0
m=audio 10010 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 192.168.0.77:6855 (NAT)
Using INVITE request as basis request - y83R9V6b3DPMSEkcREVCTg..
Found peer '200' for '200' from 192.168.0.77:6855

<--- Reliably Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>;tag=as57a67be0
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 1 INVITE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="395a17d9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'y83R9V6b3DPMSEkcREVCTg..' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.77:6855 --->
ACK sip:************@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;rport
Max-Forwards: 70
To: <sip:************@192.168.0.68>;tag=as57a67be0
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.77:6855 --->
INVITE sip:************@192.168.0.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;rport
Max-Forwards: 70
Contact: <sip:200@192.168.0.77:6855>;+sip.instance="<urn:uuid:************>"
To: <sip:************@192.168.0.68>
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: PortGo v8.1, Build 01220214
Authorization: Digest username="200",realm="asterisk",nonce="395a17d9",uri="sip:************@192.168.0.68",response="933864390e2354e7934178cc0d5a3dd5",algorithm=MD5
Content-Length: 241

v=0
o=- 8314748 8314748 IN IP4 192.168.0.77
s=http://www.portsip.com
c=IN IP4 192.168.0.77
t=0 0
m=audio 10010 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.77:6855 (no NAT)
Using INVITE request as basis request - y83R9V6b3DPMSEkcREVCTg..
Found peer '200' for '200' from 192.168.0.77:6855
== 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 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.77:10010
Looking for ************ in from-internal (domain 192.168.0.68)
list_route: hop: <sip:200@192.168.0.77:6855>

<--- Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:************@192.168.0.68:5060>
Content-Length: 0


<------------>
-- Executing [************@from-internal:1] Macro("SIP/200-0000000e", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/200-0000000e", "TOUCH_MONITOR=1418279015.14") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/200-0000000e", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/200-0000000e", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/200-0000000e", "1?Set(REALCALLERIDNUM=200)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/200-0000000e", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/200-0000000e", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-0000000e", "AMPUSERCIDNAME=200") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/200-0000000e", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/200-0000000e", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/200-0000000e", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/200-0000000e", "CALLERID(all)="200" <200>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/200-0000000e", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/200-0000000e", "1?Set(GROUP(concurrency_limit)=200)") in new stack
-- Executing [s@macro-user-callerid:14] GosubIf("SIP/200-0000000e", "7?sub-ccss,s,1(from-internal,************)") in new stack
-- Executing [s@sub-ccss:1] ExecIf("SIP/200-0000000e", "0?Return()") in new stack
-- Executing [s@sub-ccss:2] Set("SIP/200-0000000e", "CCSS_SETUP=TRUE") in new stack
-- Executing [s@sub-ccss:3] GosubIf("SIP/200-0000000e", "0?monitor_config,1(from-internal,************):monitor_default,1(from-internal,************)") in new stack
-- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/200-0000000e", "0?is_exten") in new stack
-- Executing [monitor_default@sub-ccss:2] StackPop("SIP/200-0000000e", "") in new stack
-- Executing [monitor_default@sub-ccss:3] Return("SIP/200-0000000e", "FALSE") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("SIP/200-0000000e", "1?Set(CHANNEL(language)=Admin)") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/200-0000000e", "1?continue") in new stack
-- Goto (macro-user-callerid,s,30)
-- Executing [s@macro-user-callerid:30] Set("SIP/200-0000000e", "CALLERID(number)=200") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/200-0000000e", "CALLERID(name)=200") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/200-0000000e", "CDR(cnum)=200") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/200-0000000e", "CDR(cnam)=200") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/200-0000000e", "CHANNEL(language)=Admin") in new stack
-- Executing [************@from-internal:2] ExecIf("SIP/200-0000000e", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [************@from-internal:3] Set("SIP/200-0000000e", "MOHCLASS=default") in new stack
-- Executing [************@from-internal:4] Set("SIP/200-0000000e", "_NODEST=") in new stack
-- Executing [************@from-internal:5] Gosub("SIP/200-0000000e", "sub-record-check,s,1(out,************,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/200-0000000e", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/200-0000000e", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/200-0000000e", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/200-0000000e", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/200-0000000e", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/200-0000000e", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/200-0000000e", "0?out,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/200-0000000e", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/200-0000000e", "NOW=1418279015") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/200-0000000e", "__DAY=11") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/200-0000000e", "__MONTH=12") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/200-0000000e", "__YEAR=2014") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/200-0000000e", "__TIMESTR=20141211-092335") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/200-0000000e", "__FROMEXTEN=200") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/200-0000000e", "__CALLFILENAME=out-************-200-20141211-092335-1418279015.14") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/200-0000000e", "out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] ExecIf("SIP/200-0000000e", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
-- Executing [out@sub-record-check:2] GosubIf("SIP/200-0000000e", "0?record,1(exten,************,200)") in new stack
-- Executing [out@sub-record-check:3] Return("SIP/200-0000000e", "") in new stack
-- Executing [************@from-internal:6] Macro("SIP/200-0000000e", "dialout-trunk,2,************,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/200-0000000e", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/200-0000000e", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/200-0000000e", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/200-0000000e", "DIAL_NUMBER=************") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/200-0000000e", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/200-0000000e", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/200-0000000e", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/200-0000000e", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/200-0000000e", "DIAL_TRUNK_OPTIONS=Tt") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/200-0000000e", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/200-0000000e", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/200-0000000e", "0?Set(REALCALLERIDNUM=200)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/200-0000000e", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/200-0000000e", "USEROUTCID=200") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/200-0000000e", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/200-0000000e", "TRUNKOUTCID=************") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/200-0000000e", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/200-0000000e", "1?Set(CALLERID(all)=************)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/200-0000000e", "1?Set(CALLERID(all)=200)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/200-0000000e", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/200-0000000e", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/200-0000000e", "CDR(outbound_cnum)=200") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/200-0000000e", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/200-0000000e", "0?sub-flp-2,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/200-0000000e", "OUTNUM=************") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/200-0000000e", "custom=SIP/************") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/200-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/200-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/200-0000000e", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/200-0000000e", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/200-0000000e", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/200-0000000e", "1?Set(CONNECTEDLINE(num,i)=************)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/200-0000000e", "1?Set(CONNECTEDLINE(name,i)=CID:200)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/200-0000000e", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/200-0000000e", "SIP/************/************,300,Tt") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/************/************
-- SIP/************-0000000f is ringing

<--- Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>;tag=as5593437b
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:************@192.168.0.68:5060>
P-Asserted-Identity: "CID:200" <sip:************@192.168.0.68>
Content-Length: 0
<------------>
-- SIP/************-0000000f is making progress passing it to SIP/200-0000000e
Audio is at 10030
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>;tag=as5593437b
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:************@192.168.0.68:5060>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1826738548 1826738548 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.32.1
c=IN IP4 192.168.0.68
t=0 0
m=audio 10030 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.0.77:6855 --->


<------------->
Reliably Transmitting (no NAT) to 192.168.0.77:6855:
OPTIONS sip:200@192.168.0.77:6855 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=*****************
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.68>;tag=as25104baa
To: <sip:200@192.168.0.77:6855>
Contact: <sip:Unknown@192.168.0.68:5060>
Call-ID: 113cc51639042ba20c53aa1a23b4aae7@192.168.0.68:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.32.1)
Date: Thu, 11 Dec 2014 06:23:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.77:6855 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.68:5060;branch=************
Contact: <sip:192.168.0.77:6855>
To: <sip:200@192.168.0.77:6855>;tag=d42a2f0d
From: "Unknown" <sip:Unknown@192.168.0.68>;tag=as25104baa
Call-ID: 113cc51639042ba20c53aa1a23b4aae7@192.168.0.68:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO
Supported: replaces
User-Agent: PortGo v8.1, Build 01220214
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '113cc51639042ba20c53aa1a23b4aae7@192.168.0.68:5060' Method: OPTIONS
-- SIP/************-0000000f is ringing
-- SIP/************-0000000f answered SIP/200-0000000e
Audio is at 10030
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>;tag=as5593437b
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:************@192.168.0.68:5060>
P-Asserted-Identity: "CID:200" <sip:************@192.168.0.68>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1826738548 1826738548 IN IP4 192.168.0.68
s=Asterisk PBX 1.8.32.1
c=IN IP4 192.168.0.68
t=0 0
m=audio 10030 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.0.77:6855 --->
ACK sip:************@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=z9hG4bK-524287-1---f178af3aa6648633;rport
Max-Forwards: 70
Contact: <sip:200@192.168.0.77:6855>;+sip.instance="<urn:uuid:************>"
To: <sip:************@192.168.0.68>;tag=as5593437b
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 2 ACK
User-Agent: PortGo v8.1, Build 01220214
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.77:6855 --->
BYE sip:************@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;rport
Max-Forwards: 70
Contact: <sip:200@192.168.0.77:6855>;+sip.instance="<urn:uuid:************>"
To: <sip:************@192.168.0.68>;tag=as5593437b
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 3 BYE
User-Agent: PortGo v8.1, Build 01220214
Authorization: Digest username="200",realm="asterisk",nonce="395a17d9",uri="sip:************@192.168.0.68:5060",response="7e76640fa6080e2684e963ecc31f5fba",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.77:6855 (no NAT)
Scheduling destruction of SIP dialog 'y83R9V6b3DPMSEkcREVCTg..' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.77:6855 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.77:6855;branch=************;received=192.168.0.77;rport=6855
From: "Sergeevav"<sip:200@192.168.0.68>;tag=a935c171
To: <sip:************@192.168.0.68>;tag=as5593437b
Call-ID: y83R9V6b3DPMSEkcREVCTg..
CSeq: 3 BYE
Server: FPBX-2.11.0(1.8.32.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/200-0000000e", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/200-0000000e", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/200-0000000e", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/200-0000000e", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/200-0000000e' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/200-0000000e'
== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/200-0000000e' in macro 'dialout-trunk'
== Spawn extension (from-internal, ************, 6) exited non-zero on 'SIP/200-0000000e'
И проблема только с этим провайдером. Звонки через других проходят нормально.

Подскажите пожалуйста в куда копать?
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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: FreePBX и исходящие

Сообщение ded »

1) А почему техподдержка провайдера это не раскуривает?
2) Подскажите, в чём смысл скрывать внутренний адрес 10.*.*. провайдерского прокси?
Кто-то прочитает, и побежит хакать адрес 10.*.*??

Укажите
localnet=10.0.0.0/255.0.0.0
Это и есть информация для RTP при соединении - не проксировать.
НО возможно будет проблема ещё глубже. Если есть прямой маршрут на 10.*.* (проверьте через mtr) то возможно провайдерский прокси готов принимать RTP только с вашего интерфейса с адресом 10.*.*.
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sergeev2d
Сообщения: 3
Зарегистрирован: 24 ноя 2014, 11:04

Re: FreePBX и исходящие

Сообщение sergeev2d »

1) Тех поддержка шлет лесом и говорит что у них все ок, а у меня проблема с кодеками.
2) А я что то не знаю) найти и заменить, навязчивая идея.

Указал в custom новый localnet. Проблема осталась.
mtr не достает проксю эту.
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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: FreePBX и исходящие

Сообщение ded »

1) посылайте этого оператора лесом, голосуйте ногами, так сказать. Переходите к нам, мы тоже сам себе режиссёр, лесом клиентов не посылаем. А оказывать бесплатную поддержку для платных чужих операторов - влом. Поэтому -
2) Платный суппорт.
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sergeev2d
Сообщения: 3
Зарегистрирован: 24 ноя 2014, 11:04

Re: FreePBX и исходящие

Сообщение sergeev2d »

ded писал(а):1) посылайте этого оператора лесом, голосуйте ногами, так сказать. Переходите к нам, мы тоже сам себе режиссёр, лесом клиентов не посылаем. А оказывать бесплатную поддержку для платных чужих операторов - влом. Поэтому -
2) Платный суппорт.
Спасибо за помощь
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