проблема с исходящим вызовом

[merkajiu]

Добрый день!

Имеется транк с провайдером. Входящие звонки работают. Но исходящих нет. Провайдер говорит

некорректно формируется поле TO и Request-URI

Request-Line: INVITE sip:+77123456789%405.63.114.58@провайдер SIP/2.0

To: sip:+77123456789%405.63.114.58@провайдер

Должно быть +77123456789@провайдер

не понятно откуда взялось "%405.63.114.58"

Подскажите куда копать

Freepbx

[Kroteg]

Настройки транка с провайдером или строка регистрации. Телепаты в отпуске.
Возможно это поле fromdomain или правда из строки регистрации берется.

[merkajiu]

Извиняюсь.
вот параметры настройки с транкам провайдера

Freepbx
trunk
Outgoing

type=peer
qualify=60000
host=195.111.11.119
disallow=all
allow=alaw&g729

Провайдер говорит что появилось после добавления useragent=Igroup
useragent добавил чтобы транк работал. Иначе не работает с скобками useragent=Freepbx 13 (13.01)

[ded]

В параметры транка добавьте
fromdomain=195.111.11.119

строка Dial в консоли должна выглядеть как
Dial(SIP/trunkname/79012345678,,t)
а не Dial(SIP/79012345678@trunkname,,t)

P.S. qualify=60000 - зачем это? Кто посоветовал? Это глупость.

[Kroteg]

не хватает:
directmedia=
context=from-pstn
dtmfmode=
username=
secret=

по желанию:
fromdomain=
fromuser=
insecure=port,invite
outboundproxy=
port=

qualify разве не в секундах задается? и как правило достаточно просто yes

[ded]

Параметры
insecure=port,invite
context=from-pstn
нужны для входящих, а ТС пишет о проблемах исходящих.

[merkajiu]

Здравствуйте!
добавил fromdomain.
Без изменений
Два интерфейса 192.168.1.113 и белая 5.36.214.85
route provider_ip via 5.36.214.86
Прикрепил лог

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

freepbx*CLI> sip set debug peer 101
SIP Debugging Enabled for IP: 192.168.1.51

<--- SIP read from UDP:192.168.1.51:5060 --->
INVITE sip:+77123456789@192.168.1.113 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118747a91cf6d3c9ca;rport
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 44 INVITE
Contact: <sip:101@192.168.1.51:5060;gr=80AA0C04-73BC-E811-871B-A91CF6D3C9CA>
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, gruu, timer
P-Preferred-Identity: <sip:101@192.168.1.113>
Content-Length: 547

v=0
o=- 2573063209 1 IN IP4 192.168.1.51
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.51
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101
a=rtpmap:107 opus/48000/2
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:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:11 L16/44100
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3651402496
a=sendrecv
<------------->
--- (17 headers 23 lines) ---
Sending to 192.168.1.51:5060 (NAT)
Sending to 192.168.1.51:5060 (NAT)
Using INVITE request as basis request - 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
Found peer '101' for '101' from 192.168.1.51:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.51:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118747a91cf6d3c9ca;received=192.168.1.51;rport=5060
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>;tag=as0aa9d3bb
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 44 INVITE
Server: DibaGroup
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0778ff4b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.51:5060 --->
ACK sip:+77123456789@192.168.1.113 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118747a91cf6d3c9ca;rport
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>;tag=as0aa9d3bb
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 44 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.51:5060 --->
INVITE sip:+77123456789@192.168.1.113 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118748a91cf6d3c9ca;rport
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 45 INVITE
Contact: <sip:101@192.168.1.51:5060;gr=80AA0C04-73BC-E811-871B-A91CF6D3C9CA>
Authorization: Digest username="101", realm="asterisk", nonce="0778ff4b", uri="sip:+77123456789@192.168.1.113", response="565ff26571b3905bdbab3c48afe8fede", algorithm=MD5
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, gruu, timer
P-Preferred-Identity: <sip:101@192.168.1.113>
Content-Length: 547

v=0
o=- 2573063209 1 IN IP4 192.168.1.51
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.51
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101
a=rtpmap:107 opus/48000/2
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:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:11 L16/44100
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3651402496
a=sendrecv
<------------->
--- (18 headers 23 lines) ---
Sending to 192.168.1.51:5060 (no NAT)
Using INVITE request as basis request - 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
Found peer '101' for '101' from 192.168.1.51:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 107
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 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 11
Found RTP audio format 118
Found RTP audio format 101
Found audio description format opus for ID 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 speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found unknown media description format L16 for ID 11
Found audio description format L16 for ID 118
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|g729|ilbc|opus|speex|speex16|slin16)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7fe43c034470 -- Strict RTP learning after remote address set to: 192.168.1.51:5062
Peer audio RTP is at port 192.168.1.51:5062
Looking for +77123456789 in from-internal (domain 192.168.1.113)
sip_route_dump: route/path hop: <sip:101@192.168.1.51:5060;gr=80AA0C04-73BC-E811-871B-A91CF6D3C9CA>

<--- Transmitting (no NAT) to 192.168.1.51:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118748a91cf6d3c9ca;received=192.168.1.51;rport=5060
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 45 INVITE
Server: DibaGroup
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+77123456789@192.168.1.113:5060>
Content-Length: 0


<------------>
-- Executing [+77123456789@from-internal:1] Macro("SIP/101-000006b7", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/101-000006b7", "TOUCH_MONITOR=1537585439.4805") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/101-000006b7", "AMPUSER=101") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/101-000006b7", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/101-000006b7", "1?Set(REALCALLERIDNUM=101)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/101-000006b7", "AMPUSER=101") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-000006b7", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/101-000006b7", "AMPUSERCIDNAME=101") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("SIP/101-000006b7", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/101-000006b7", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/101-000006b7", "AMPUSERCID=101") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/101-000006b7", "__DIAL_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/101-000006b7", "CALLERID(all)="101" <101>") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/101-000006b7", "0?limit") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/101-000006b7", "1?Set(GROUP(concurrency_limit)=101)") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("SIP/101-000006b7", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:16] NoOp("SIP/101-000006b7", "Macro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/101-000006b7", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] GotoIf("SIP/101-000006b7", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [s@macro-user-callerid:37] Set("SIP/101-000006b7", "CALLERID(number)=101") in new stack
-- Executing [s@macro-user-callerid:38] Set("SIP/101-000006b7", "CALLERID(name)=101") in new stack
-- Executing [s@macro-user-callerid:39] GotoIf("SIP/101-000006b7", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:40] Set("SIP/101-000006b7", "CDR(cnam)=101") in new stack
-- Executing [s@macro-user-callerid:41] Set("SIP/101-000006b7", "CDR(cnum)=101") in new stack
-- Executing [s@macro-user-callerid:42] Set("SIP/101-000006b7", "CHANNEL(language)=ru") in new stack
-- Executing [+77123456789@from-internal:2] Gosub("SIP/101-000006b7", "sub-record-check,s,1(out,+77123456789,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/101-000006b7", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/101-000006b7", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/101-000006b7", "NOW=1537585439") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/101-000006b7", "__DAY=22") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/101-000006b7", "__MONTH=09") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/101-000006b7", "__YEAR=2018") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/101-000006b7", "__TIMESTR=20180922-030359") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/101-000006b7", "__FROMEXTEN=101") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/101-000006b7", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/101-000006b7", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/101-000006b7", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/101-000006b7", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/101-000006b7", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/101-000006b7", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/101-000006b7", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/101-000006b7", "Outbound Recording Check from 101 to +77123456789") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/101-000006b7", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/101-000006b7", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/101-000006b7", "recordcheck,1(dontcare,out,+77123456789)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/101-000006b7", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/101-000006b7", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/101-000006b7", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/101-000006b7", "") in new stack
-- Executing [+77123456789@from-internal:3] ExecIf("SIP/101-000006b7", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [+77123456789@from-internal:4] Set("SIP/101-000006b7", "MOHCLASS=default") in new stack
-- Executing [+77123456789@from-internal:5] Set("SIP/101-000006b7", "_NODEST=") in new stack
-- Executing [+77123456789@from-internal:6] Macro("SIP/101-000006b7", "dialout-trunk,1,+77123456789,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/101-000006b7", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-000006b7", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] ExecIf("SIP/101-000006b7", "0?Set(CALLERID(num)=101)") in new stack
-- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/101-000006b7", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/101-000006b7", "DIAL_NUMBER=+77123456789") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/101-000006b7", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("SIP/101-000006b7", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("SIP/101-000006b7", "DIAL_TRUNK_OPTIONS=T") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-000006b7", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,11)
-- Executing [s@macro-dialout-trunk:11] GotoIf("SIP/101-000006b7", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:12] Macro("SIP/101-000006b7", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] NoOp("SIP/101-000006b7", "101") in new stack
-- Executing [s@macro-outbound-callerid:2] NoOp("SIP/101-000006b7", "") in new stack
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/101-000006b7", "on") in new stack
-- Executing [s@macro-outbound-callerid:4] ExecIf("SIP/101-000006b7", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:5] ExecIf("SIP/101-000006b7", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:6] ExecIf("SIP/101-000006b7", "0?Set(REALCALLERIDNUM=101)") in new stack
-- Executing [s@macro-outbound-callerid:7] GotoIf("SIP/101-000006b7", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing [s@macro-outbound-callerid:11] Set("SIP/101-000006b7", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:12] Set("SIP/101-000006b7", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:13] Set("SIP/101-000006b7", "TRUNKOUTCID=+77780468886") in new stack
-- Executing [s@macro-outbound-callerid:14] GotoIf("SIP/101-000006b7", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,19)
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/101-000006b7", "1?Set(CALLERID(all)=+77780468886)") in new stack
-- Executing [s@macro-outbound-callerid:20] ExecIf("SIP/101-000006b7", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:21] ExecIf("SIP/101-000006b7", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:22] ExecIf("SIP/101-000006b7", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:23] ExecIf("SIP/101-000006b7", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:24] Set("SIP/101-000006b7", "CDR(outbound_cnum)=+77780468886") in new stack
-- Executing [s@macro-outbound-callerid:25] Set("SIP/101-000006b7", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:13] GosubIf("SIP/101-000006b7", "0?sub-flp-1,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/101-000006b7", "OUTNUM=+77123456789") in new stack
-- Executing [s@macro-dialout-trunk:15] Set("SIP/101-000006b7", "custom=SIP/out") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/101-000006b7", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
-- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/101-000006b7", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:18] Macro("SIP/101-000006b7", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-000006b7", "") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/101-000006b7", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:20] Set("SIP/101-000006b7", "__CRM_DIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("SIP/101-000006b7", "__CRM_DESTINATION=+77123456789") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("SIP/101-000006b7", "__CRM_SOURCE=101") in new stack
-- Executing [s@macro-dialout-trunk:23] AGI("SIP/101-000006b7", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/101-000006b7>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:24] Set("SIP/101-000006b7", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:25] NoOp("SIP/101-000006b7", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:26] GotoIf("SIP/101-000006b7", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/101-000006b7", "1?Set(CONNECTEDLINE(num,i)=+77123456789)") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/101-000006b7", "1?Set(CONNECTEDLINE(name,i)=CID:+77780468886)") in new stack
-- Executing [s@macro-dialout-trunk:29] ExecIf("SIP/101-000006b7", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)+77780468886)") in new stack
-- Executing [s@macro-dialout-trunk:30] GotoIf("SIP/101-000006b7", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:31] Dial("SIP/101-000006b7", "SIP/out/+77123456789@5.36.214.85,300,T") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/out/+77123456789@5.36.214.85
[2018-09-22 03:03:59] WARNING[11388][C-00000196]: chan_sip.c:24071 handle_response_invite: Received response: "Forbidden" from '<sip:+77780468886@195.47.255.119>;tag=as32aa2f00'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:32] NoOp("SIP/101-000006b7", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:33] GotoIf("SIP/101-000006b7", "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/101-000006b7", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-000006b7", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/101-000006b7", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/101-000006b7", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/101-000006b7", "1?Set(CALLERID(number)=101)") in new stack
-- Executing [+77123456789@from-internal:7] Macro("SIP/101-000006b7", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/101-000006b7", "") in new stack
Audio is at 16586
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.1.51:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118748a91cf6d3c9ca;received=192.168.1.51;rport=5060
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>;tag=as5e96377e
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 45 INVITE
Server: DibaGroup
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+77123456789@192.168.1.113:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 324

v=0
o=root 1630349521 1630349521 IN IP4 192.168.1.113
s=Asterisk PBX 14.7.4
c=IN IP4 192.168.1.113
t=0 0
m=audio 16586 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/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/101-000006b7", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-000006b7", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/101-000006b7", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
-- <SIP/101-000006b7> Playing 'all-circuits-busy-now.ulaw' (language 'ru')
> 0x7fe43c034470 -- Strict RTP switching to RTP target address 192.168.1.51:5062 as source
> 0x7fe43c034470 -- Strict RTP learning complete - Locking on source address 192.168.1.51:5062
-- <SIP/101-000006b7> Playing 'please-try-call-later.ulaw' (language 'ru')
Reliably Transmitting (no NAT) to 192.168.1.51:5060:
OPTIONS sip:101@192.168.1.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK1753e15f
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.113>;tag=as68a51356
To: <sip:101@192.168.1.51:5060>
Contact: <sip:Unknown@192.168.1.113:5060>
Call-ID: 583223b95dda98417e243041509b552b@192.168.1.113:5060
CSeq: 102 OPTIONS
User-Agent: DibaGroup
Date: Sat, 22 Sep 2018 03:04:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK1753e15f
From: "Unknown" <sip:Unknown@192.168.1.113>;tag=as68a51356
To: <sip:101@192.168.1.51:5060>;tag=802d7bd681bce8118748a91cf6d3c9ca
Call-ID: 583223b95dda98417e243041509b552b@192.168.1.113:5060
CSeq: 102 OPTIONS
Contact: <sip:101@192.168.1.51:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '583223b95dda98417e243041509b552b@192.168.1.113:5060' Method: OPTIONS
-- Executing [s@macro-outisbusy:5] Congestion("SIP/101-000006b7", "20") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.51:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118748a91cf6d3c9ca;received=192.168.1.51;rport=5060
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>;tag=as5e96377e
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 45 INVITE
Server: DibaGroup
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
[2018-09-22 03:04:04] WARNING[28978][C-00000196]: channel.c:5005 ast_prod: Prodding channel 'SIP/101-000006b7' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-000006b7' in macro 'outisbusy'
== Spawn extension (from-internal, +77123456789, 7) exited non-zero on 'SIP/101-000006b7'
-- Executing [h@from-internal:1] Macro("SIP/101-000006b7", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-000006b7", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/101-000006b7", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("SIP/101-000006b7", " monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("SIP/101-000006b7", "attendedtransfer-rec-restart.php,,") in new stack

<--- SIP read from UDP:192.168.1.51:5060 --->
ACK sip:+77123456789@192.168.1.113 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK80d318d481bce8118748a91cf6d3c9ca;rport
From: "PhonerLite" <sip:101@192.168.1.113>;tag=2779755399
To: <sip:+77123456789@192.168.1.113>;tag=as5e96377e
Call-ID: 80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51
CSeq: 45 ACK
Authorization: Digest username="101", realm="asterisk", nonce="0778ff4b", uri="sip:+77123456789@192.168.1.113", response="565ff26571b3905bdbab3c48afe8fede", algorithm=MD5
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <SIP/101-000006b7>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("SIP/101-000006b7", "") in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/101-000006b7' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-000006b7'
-- SIP/101-000006b7 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/101-000006b7", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/101-000006b7", "HANGUP CAUSE: 34") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/101-000006b7", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/101-000006b7", "MASTER CHANNEL: 1537585439.4805 = 1537585439.4805") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/101-000006b7", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/101-000006b7", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/101-000006b7", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/101-000006b7>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/101-000006b7", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-000006b7'
-- SIP/101-000006b7 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
Really destroying SIP dialog '80D318D4-81BC-E811-8746-A91CF6D3C9CA@192.168.1.51' Method: ACK
freepbx*CLI> sip set debug off
SIP Debugging Disabled
freepbx*CLI> exit

[merkajiu]

tcpdump

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

No. Time Source Destination Protocol Length Info
2 0.826563 5.36.214.85 195.11.111.119 SIP/SDP 944 Request: INVITE sip:+77123456789%405.36.214.85@195.11.111.119 |

Frame 2: 944 bytes on wire (7552 bits), 944 bytes captured (7552 bits)
Ethernet II, Src: AsustekC_70:13:99 (2c:fd:a1:70:13:99), Dst: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0)
Internet Protocol Version 4, Src: 5.36.214.85, Dst: 195.11.111.119
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)

No. Time Source Destination Protocol Length Info
3 0.852646 195.11.111.119 5.36.214.85 SIP 338 Status: 100 Trying |

Frame 3: 338 bytes on wire (2704 bits), 338 bytes captured (2704 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 195.11.111.119, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (100)

No. Time Source Destination Protocol Length Info
4 0.872481 195.11.111.119 5.36.214.85 SIP 414 Status: 403 Forbidden |

Frame 4: 414 bytes on wire (3312 bits), 414 bytes captured (3312 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 195.11.111.119, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (403)

No. Time Source Destination Protocol Length Info
5 0.872535 5.36.214.85 195.11.111.119 SIP 483 Request: ACK sip:+77123456789%405.36.214.85@195.11.111.119 |

Frame 5: 483 bytes on wire (3864 bits), 483 bytes captured (3864 bits)
Ethernet II, Src: AsustekC_70:13:99 (2c:fd:a1:70:13:99), Dst: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0)
Internet Protocol Version 4, Src: 5.36.214.85, Dst: 195.11.111.119
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (ACK)

No. Time Source Destination Protocol Length Info
6 2.969933 178.90.90.57 5.36.214.85 SIP 611 Request: REGISTER sip:5.36.214.85;transport=UDP (1 binding) |

Frame 6: 611 bytes on wire (4888 bits), 611 bytes captured (4888 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 178.90.90.57, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 40445, Dst Port: 5060
Session Initiation Protocol (REGISTER)

No. Time Source Destination Protocol Length Info
7 3.463803 178.90.90.57 5.36.214.85 SIP 611 Request: REGISTER sip:5.36.214.85;transport=UDP (1 binding) |

Frame 7: 611 bytes on wire (4888 bits), 611 bytes captured (4888 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 178.90.90.57, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 40445, Dst Port: 5060
Session Initiation Protocol (REGISTER)

No. Time Source Destination Protocol Length Info
8 4.464693 178.90.90.57 5.36.214.85 SIP 611 Request: REGISTER sip:5.36.214.85;transport=UDP (1 binding) |

Frame 8: 611 bytes on wire (4888 bits), 611 bytes captured (4888 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 178.90.90.57, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 40445, Dst Port: 5060
Session Initiation Protocol (REGISTER)

No. Time Source Destination Protocol Length Info
9 6.465734 178.90.90.57 5.36.214.85 SIP 611 Request: REGISTER sip:5.36.214.85;transport=UDP (1 binding) |

Frame 9: 611 bytes on wire (4888 bits), 611 bytes captured (4888 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 178.90.90.57, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 40445, Dst Port: 5060
Session Initiation Protocol (REGISTER)

No. Time Source Destination Protocol Length Info
10 10.466195 178.90.90.57 5.36.214.85 SIP 611 Request: REGISTER sip:5.36.214.85;transport=UDP (1 binding) |

Frame 10: 611 bytes on wire (4888 bits), 611 bytes captured (4888 bits)
Ethernet II, Src: Cisco_ee:12:c0 (00:1b:0d:ee:12:c0), Dst: AsustekC_70:13:99 (2c:fd:a1:70:13:99)
Internet Protocol Version 4, Src: 178.90.90.57, Dst: 5.36.214.85
User Datagram Protocol, Src Port: 40445, Dst Port: 5060
Session Initiation Protocol (REGISTER)

[awsswa]

tcpdump ?
Серьезно ?
Спустились с гор за спичками и сразу настраивать ?
Неужто трудно сделать пару запросов в гугле и посмотреть как выглядит пакет INVITE ?

вот что должны прислать
https://voipnotes.ru/change-field-user-agent-asterisk/

[merkajiu]

Здравствуйте!
Добавил в соответствии с статьей.
исходящие звонки не доступны.