Пропадает связь через 30 сек

freeman33 · Сообщение **freeman33** » 27 май 2021, 11:54

Zavr2008 писал(а):
разговор между двумя абонентами в течение 30 секунд это не показатель
Логи консоли астера видите? там же ясно наверняка пишет про Critical packet retransmission..
30 секунд - стандартный таймер в SIP.

вот лог звонка, не нашел ни одного слова из фраы Critical packet retransmission
я понимаю что вы тут все специалисты и диагностировали бы проблему за 5 мин., но я затем сюда и обратился чтоб разъяснили в чем дело. Связь есть, рвзговор есть, дело за малым осталось, понять почему отключает.
Потом уже буду вторую сторону копать- звонки с asterisk'a на freeswitch.
Какой момент должен писать Asterisk про Critical packet retransmission ?

core set verbose 10

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
<--- Received SIP request (1300 bytes) from UDP:172.18.253.1:5060 --->
INVITE sip:180@192.168.103.3 SIP/2.0
Via: SIP/2.0/UDP 172.18.253.1;rport;branch=z9hG4bKSjyBecUBXHcSr
Max-Forwards: 69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
CSeq: 36498768 INVITE
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 315
X-accountcode: 172.18.253.1
X-FS-Support: update_display,send_info
Remote-Party-ID: "2788280" <sip:2788280@uraltep>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1622093175 1622093176 IN IP4 172.18.253.1
s=FreeSWITCH
c=IN IP4 172.18.253.1
t=0 0
m=audio 30142 RTP/AVP 9 0 8 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

<--- Transmitting SIP response (502 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKSjyBecUBXHcSr
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=z9hG4bKSjyBecUBXHcSr
CSeq: 36498768 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1622105121/f78ecbc0fb1d12b57f46b8187d780954",opaque="335003386620c158",algorithm=md5,qop="auth"
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

<--- Received SIP request (318 bytes) from UDP:172.18.253.1:5060 --->
ACK sip:180@192.168.103.3 SIP/2.0
Via: SIP/2.0/UDP 172.18.253.1;rport;branch=z9hG4bKSjyBecUBXHcSr
Max-Forwards: 69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=z9hG4bKSjyBecUBXHcSr
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
CSeq: 36498768 ACK
Content-Length: 0

<--- Received SIP request (1586 bytes) from UDP:172.18.253.1:5060 --->
INVITE sip:180@192.168.103.3 SIP/2.0
Via: SIP/2.0/UDP 172.18.253.1;rport;branch=z9hG4bKtUQ4F7BFtt2Bm
Max-Forwards: 69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
CSeq: 36498769 INVITE
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Authorization: Digest username="ekttrank", realm="asterisk", nonce="1622105121/f78ecbc0fb1d12b57f46b8187d780954", cnonce="tqJw3zlqEjqPrQAMKcHaaQ", opaque="335003386620c158", algorithm=MD5, uri="sip:180@192.168.103.3", response="b3a177083152af894fae935ac2abd199", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 315
X-accountcode: 172.18.253.1
X-FS-Support: update_display,send_info
Remote-Party-ID: "2788280" <sip:2788280@uraltep>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1622093175 1622093176 IN IP4 172.18.253.1
s=FreeSWITCH
c=IN IP4 172.18.253.1
t=0 0
m=audio 30142 RTP/AVP 9 0 8 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

== Setting global variable 'SIPDOMAIN' to '192.168.103.3'
<--- Transmitting SIP response (325 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

-- Executing [180@from-pstn:1] GotoIf("PJSIP/ekttrank-0000006f", "1?ext-local,180,1:followme-check,180,1") in new stack
-- Goto (ext-local,180,1)
-- Executing [180@ext-local:1] Set("PJSIP/ekttrank-0000006f", "__RINGTIMER=15") in new stack
-- Executing [180@ext-local:2] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(__CWIGNORE=)") in new stack
-- Executing [180@ext-local:3] Macro("PJSIP/ekttrank-0000006f", "exten-vm,novm,180,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("PJSIP/ekttrank-0000006f", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/ekttrank-0000006f", "TOUCH_MONITOR=1622105121.113") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/ekttrank-0000006f", "AMPUSER=ekttrank") in new stack
-- Executing [s@macro-user-callerid:3] Set("PJSIP/ekttrank-0000006f", "HOTDESCKCHAN=ekttrank-0000006f") in new stack
-- Executing [s@macro-user-callerid:4] Set("PJSIP/ekttrank-0000006f", "HOTDESKEXTEN=ekttrank") in new stack
-- Executing [s@macro-user-callerid:5] Set("PJSIP/ekttrank-0000006f", "HOTDESKCALL=0") in new stack
-- Executing [s@macro-user-callerid:6] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(HOTDESKCALL=1)") in new stack
-- Executing [s@macro-user-callerid:7] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CALLERID(name)=)") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/ekttrank-0000006f", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("PJSIP/ekttrank-0000006f", "1?Set(REALCALLERIDNUM=ekttrank)") in new stack
-- Executing [s@macro-user-callerid:10] Set("PJSIP/ekttrank-0000006f", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("PJSIP/ekttrank-0000006f", "0?limit") in new stack
-- Executing [s@macro-user-callerid:12] Set("PJSIP/ekttrank-0000006f", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/ekttrank-0000006f", "1?report") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("PJSIP/ekttrank-0000006f", "Macro Depth is 2") in new stack
-- Executing [s@macro-user-callerid:24] GotoIf("PJSIP/ekttrank-0000006f", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,25)
-- Executing [s@macro-user-callerid:25] GotoIf("PJSIP/ekttrank-0000006f", "0?continue") in new stack
-- Executing [s@macro-user-callerid:26] ExecIf("PJSIP/ekttrank-0000006f", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [s@macro-user-callerid:27] Set("PJSIP/ekttrank-0000006f", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:28] GotoIf("PJSIP/ekttrank-0000006f", "1?continue") in new stack
-- Goto (macro-user-callerid,s,44)
-- Executing [s@macro-user-callerid:44] Set("PJSIP/ekttrank-0000006f", "CALLERID(number)=ekttrank") in new stack
-- Executing [s@macro-user-callerid:45] Set("PJSIP/ekttrank-0000006f", "CALLERID(name)=2788280") in new stack
-- Executing [s@macro-user-callerid:46] GotoIf("PJSIP/ekttrank-0000006f", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:47] Set("PJSIP/ekttrank-0000006f", "CDR(cnam)=2788280") in new stack
-- Executing [s@macro-user-callerid:48] Set("PJSIP/ekttrank-0000006f", "CDR(cnum)=ekttrank") in new stack
-- Executing [s@macro-user-callerid:49] Set("PJSIP/ekttrank-0000006f", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-exten-vm:2] Set("PJSIP/ekttrank-0000006f", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("PJSIP/ekttrank-0000006f", "__EXTTOCALL=180") in new stack
-- Executing [s@macro-exten-vm:4] Set("PJSIP/ekttrank-0000006f", "__PICKUPMARK=180") in new stack
-- Executing [s@macro-exten-vm:5] Set("PJSIP/ekttrank-0000006f", "RT=") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:6] ExecIf("PJSIP/ekttrank-0000006f", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:7] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:8] ExecIf("PJSIP/ekttrank-0000006f", "0?Gosub(ext-intercom,*80180,1())") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:9] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:10] ExecIf("PJSIP/ekttrank-0000006f", "0?ChanSpy(PJSIP/180,q)") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:11] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
[2021-05-27 15:45:21] WARNING[10298][C-0000003b]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
-- Executing [s@macro-exten-vm:12] ExecIf("PJSIP/ekttrank-0000006f", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
-- Executing [s@macro-exten-vm:13] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:14] ExecIf("PJSIP/ekttrank-0000006f", "0?Gosub(ext-intercom,*80180,1())") in new stack
-- Executing [s@macro-exten-vm:15] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:16] ExecIf("PJSIP/ekttrank-0000006f", "0?ChanSpy(PJSIP/180,q)") in new stack
-- Executing [s@macro-exten-vm:17] ExecIf("PJSIP/ekttrank-0000006f", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:18] Gosub("PJSIP/ekttrank-0000006f", "sub-record-check,s,1(exten,180,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("PJSIP/ekttrank-0000006f", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("PJSIP/ekttrank-0000006f", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("PJSIP/ekttrank-0000006f", "NOW=1622105121") in new stack
-- Executing [s@sub-record-check:4] Set("PJSIP/ekttrank-0000006f", "__DAY=27") in new stack
-- Executing [s@sub-record-check:5] Set("PJSIP/ekttrank-0000006f", "__MONTH=05") in new stack
-- Executing [s@sub-record-check:6] Set("PJSIP/ekttrank-0000006f", "__YEAR=2021") in new stack
-- Executing [s@sub-record-check:7] Set("PJSIP/ekttrank-0000006f", "__TIMESTR=20210527-154521") in new stack
-- Executing [s@sub-record-check:8] Set("PJSIP/ekttrank-0000006f", "__FROMEXTEN=ekttrank") in new stack
-- Executing [s@sub-record-check:9] Set("PJSIP/ekttrank-0000006f", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("PJSIP/ekttrank-0000006f", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("PJSIP/ekttrank-0000006f", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("PJSIP/ekttrank-0000006f", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("PJSIP/ekttrank-0000006f", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] NoOp("PJSIP/ekttrank-0000006f", "Exten Recording Check between ekttrank and 180") in new stack
-- Executing [exten@sub-record-check:2] Set("PJSIP/ekttrank-0000006f", "CALLTYPE=internal") in new stack
-- Executing [exten@sub-record-check:3] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CALLTYPE=)") in new stack
-- Executing [exten@sub-record-check:4] Set("PJSIP/ekttrank-0000006f", "CALLEE=dontcare") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [exten@sub-record-check:6] GotoIf("PJSIP/ekttrank-0000006f", "0?callee") in new stack
-- Executing [exten@sub-record-check:7] GotoIf("PJSIP/ekttrank-0000006f", "1?caller") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [exten@sub-record-check:13] Set("PJSIP/ekttrank-0000006f", "RECMODE=") in new stack
-- Executing [exten@sub-record-check:14] Set("PJSIP/ekttrank-0000006f", "CALLERRECMODE=") in new stack
-- Executing [exten@sub-record-check:15] Set("PJSIP/ekttrank-0000006f", "CALEERECMODE=dontcare") in new stack
-- Executing [exten@sub-record-check:16] GotoIf("PJSIP/ekttrank-0000006f", "0?processnormal") in new stack
-- Executing [exten@sub-record-check:17] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:18] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:19] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:20] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:21] ExecIf("PJSIP/ekttrank-0000006f", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:22] ExecIf("PJSIP/ekttrank-0000006f", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:23] Gosub("PJSIP/ekttrank-0000006f", "recordcheck,1(dontcare,internal,180)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/ekttrank-0000006f", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/ekttrank-0000006f", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [exten@sub-record-check:24] Return("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [s@macro-exten-vm:19] GotoIf("PJSIP/ekttrank-0000006f", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,25)
-- Executing [s@macro-exten-vm:25] GosubIf("PJSIP/ekttrank-0000006f", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:26] Macro("PJSIP/ekttrank-0000006f", "dial-one,,HhTtr,180") in new stack
-- Executing [s@macro-dial-one:1] Set("PJSIP/ekttrank-0000006f", "DEXTEN=180") in new stack
-- Executing [s@macro-dial-one:2] Set("PJSIP/ekttrank-0000006f", "__CRM_SOURCE=ekttrank") in new stack
-- Executing [s@macro-dial-one:3] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(__EXTTOCALL=180)") in new stack
-- Executing [s@macro-dial-one:4] Set("PJSIP/ekttrank-0000006f", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:5] GosubIf("PJSIP/ekttrank-0000006f", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:6] GosubIf("PJSIP/ekttrank-0000006f", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:7] GotoIf("PJSIP/ekttrank-0000006f", "1?skip1") in new stack
-- Goto (macro-dial-one,s,10)
-- Executing [s@macro-dial-one:10] GotoIf("PJSIP/ekttrank-0000006f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("PJSIP/ekttrank-0000006f", "0?continue") in new stack
-- Executing [s@macro-dial-one:12] Set("PJSIP/ekttrank-0000006f", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:13] GotoIf("PJSIP/ekttrank-0000006f", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("PJSIP/ekttrank-0000006f", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,27)
-- Executing [s@macro-dial-one:27] GotoIf("PJSIP/ekttrank-0000006f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GosubIf("PJSIP/ekttrank-0000006f", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("PJSIP/ekttrank-0000006f", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("PJSIP/ekttrank-0000006f", "DEVICES=180") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("PJSIP/ekttrank-0000006f", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(DEVICES=80)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("PJSIP/ekttrank-0000006f", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("PJSIP/ekttrank-0000006f", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("PJSIP/ekttrank-0000006f", "THISDIAL=PJSIP/180") in new stack
-- Executing [dstring@macro-dial-one:8] GotoIf("PJSIP/ekttrank-0000006f", "0?docheck") in new stack
-- Executing [dstring@macro-dial-one:9] NoOp("PJSIP/ekttrank-0000006f", "Debug: Found PJSIP Destination PJSIP/180") in new stack
-- Executing [dstring@macro-dial-one:10] GotoIf("PJSIP/ekttrank-0000006f", "0?doset") in new stack
-- Executing [dstring@macro-dial-one:11] NoOp("PJSIP/ekttrank-0000006f", "Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS") in new stack
-- Executing [dstring@macro-dial-one:12] Set("PJSIP/ekttrank-0000006f", "THISDIAL=PJSIP/180/sip:180@192.168.103.198:5060") in new stack
-- Executing [dstring@macro-dial-one:13] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(DIALSTATUS=CHANUNAVAIL)") in new stack
-- Executing [dstring@macro-dial-one:14] GotoIf("PJSIP/ekttrank-0000006f", "0?skipset") in new stack
-- Executing [dstring@macro-dial-one:15] Set("PJSIP/ekttrank-0000006f", "DSTRING=PJSIP/180/sip:180@192.168.103.198:5060&") in new stack
-- Executing [dstring@macro-dial-one:16] Set("PJSIP/ekttrank-0000006f", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:17] GotoIf("PJSIP/ekttrank-0000006f", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:18] ExecIf("PJSIP/ekttrank-0000006f", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:19] Set("PJSIP/ekttrank-0000006f", "DSTRING=PJSIP/180/sip:180@192.168.103.198:5060") in new stack
-- Executing [dstring@macro-dial-one:20] Return("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [s@macro-dial-one:29] GotoIf("PJSIP/ekttrank-0000006f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:30] GotoIf("PJSIP/ekttrank-0000006f", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:31] GosubIf("PJSIP/ekttrank-0000006f", "0?ctset,1():ctclear,1()") in new stack
-- Executing [ctclear@macro-dial-one:1] NoOp("PJSIP/ekttrank-0000006f", "Deleting: CALLTRACE/180 ") in new stack
-- Executing [ctclear@macro-dial-one:2] Return("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [s@macro-dial-one:32] Set("PJSIP/ekttrank-0000006f", "D_OPTIONS=HhTtr") in new stack
-- Executing [s@macro-dial-one:33] GosubIf("PJSIP/ekttrank-0000006f", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [s@macro-dial-one:34] NoOp("PJSIP/ekttrank-0000006f", "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
-- Executing [s@macro-dial-one:35] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:36] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:37] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:38] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:39] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:40] GosubIf("PJSIP/ekttrank-0000006f", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [s@macro-dial-one:41] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:42] GosubIf("PJSIP/ekttrank-0000006f", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:43] Set("PJSIP/ekttrank-0000006f", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:44] Set("PJSIP/ekttrank-0000006f", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:45] GotoIf("PJSIP/ekttrank-0000006f", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:46] GotoIf("PJSIP/ekttrank-0000006f", "1?godial") in new stack
-- Goto (macro-dial-one,s,51)
-- Executing [s@macro-dial-one:51] Macro("PJSIP/ekttrank-0000006f", "dialout-one-predial-hook,") in new stack
-- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [s@macro-dial-one:52] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(D_OPTIONS=HhtrI)") in new stack
-- Executing [s@macro-dial-one:53] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CWRING=r(callwaiting)):Set(CWRING=)") in new stack
-- Executing [s@macro-dial-one:54] NoOp("PJSIP/ekttrank-0000006f", "") in new stack
-- Executing [s@macro-dial-one:55] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(D_OPTIONS=HhTtrg)") in new stack
-- Executing [s@macro-dial-one:56] Dial("PJSIP/ekttrank-0000006f", "PJSIP/180/sip:180@192.168.103.198:5060,,HhTtrb(func-apply-sipheaders^s^1)") in new stack
-- PJSIP/180-00000070 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/180-00000070", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/180-00000070", "Applying SIP Headers to channel PJSIP/180-00000070") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("PJSIP/180-00000070", "TECH=PJSIP") in new stack
-- Executing [s@func-apply-sipheaders:4] Set("PJSIP/180-00000070", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:5] While("PJSIP/180-00000070", "0") in new stack
-- Jumping to priority 13
-- Executing [s@func-apply-sipheaders:14] Return("PJSIP/180-00000070", "") in new stack
== Spawn extension (from-internal, 180, 1) exited non-zero on 'PJSIP/180-00000070'
-- PJSIP/180-00000070 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called PJSIP/180/sip:180@192.168.103.198:5060
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
<--- Transmitting SIP response (513 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Contact: <sip:xx.xxx.xxx.xx:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0

-- PJSIP/180-00000070 is ringing
-- PJSIP/180-00000070 is ringing
<--- Transmitting SIP response (513 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Content-Length: 0

> 0x3387ea0 -- Strict RTP learning after remote address set to: 192.168.103.198:16472
-- PJSIP/180-00000070 answered PJSIP/ekttrank-0000006f
> 0x2b2b840 -- Strict RTP learning after remote address set to: 172.18.253.1:30142
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

-- Channel PJSIP/180-00000070 joined 'simple_bridge' basic-bridge <9cbe1d08-6ef7-454f-9059-0b6f7e792216>
-- Channel PJSIP/ekttrank-0000006f joined 'simple_bridge' basic-bridge <9cbe1d08-6ef7-454f-9059-0b6f7e792216>
> 0x3387ea0 -- Strict RTP switching to RTP target address 192.168.103.198:16472 as source
> 0x2b2b840 -- Strict RTP switching to RTP target address 172.18.253.1:30142 as source
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

> 0x3387ea0 -- Strict RTP learning complete - Locking on source address 192.168.103.198:16472
> 0x2b2b840 -- Strict RTP learning complete - Locking on source address 172.18.253.1:30142
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP request (600 bytes) to UDP:172.18.253.1:5060 --->
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPjbd9d0c27-3558-4428-8d21-5274ebc77d75
From: <sip:ekttrank@192.168.103.3>;tag=fd811972-6ea3-4ac9-8c78-ec6ed988d209
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: 01885966-a58c-458b-8cb2-b06cbfcc362e
CSeq: 11697 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

<--- Received SIP response (920 bytes) from UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPjbd9d0c27-3558-4428-8d21-5274ebc77d75;received=192.168.103.3
From: <sip:ekttrank@192.168.103.3>;tag=fd811972-6ea3-4ac9-8c78-ec6ed988d209
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;tag=pFF806mZg6gNe
Call-ID: 01885966-a58c-458b-8cb2-b06cbfcc362e
CSeq: 11697 OPTIONS
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKtUQ4F7BFtt2Bm
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
To: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
CSeq: 36498769 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622093175 1622093178 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 15374 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

<--- Transmitting SIP request (495 bytes) to UDP:172.18.253.1:5060 --->
BYE sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPj91d32b99-e8ae-4988-9e6d-f6b6510bd2ce
From: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
To: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
CSeq: 21659 BYE
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

<--- Received SIP response (521 bytes) from UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPj91d32b99-e8ae-4988-9e6d-f6b6510bd2ce;received=192.168.103.3
From: <sip:180@192.168.103.3>;tag=195eeccb-9003-436a-a5c4-578c6d3f5893
To: "2788280" <sip:ekttrank@uraltep>;tag=9S6596SHpjF2Q
Call-ID: b69e7352-396a-123a-ad8f-000c29c1da69
CSeq: 21659 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0

-- Channel PJSIP/ekttrank-0000006f left 'simple_bridge' basic-bridge <9cbe1d08-6ef7-454f-9059-0b6f7e792216>
== Spawn extension (macro-dial-one, s, 56) exited non-zero on 'PJSIP/ekttrank-0000006f' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 26) exited non-zero on 'PJSIP/ekttrank-0000006f' in macro 'exten-vm'
== Spawn extension (ext-local, 180, 3) exited non-zero on 'PJSIP/ekttrank-0000006f'
-- Executing [h@ext-local:1] Macro("PJSIP/ekttrank-0000006f", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/ekttrank-0000006f", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Channel PJSIP/180-00000070 left 'simple_bridge' basic-bridge <9cbe1d08-6ef7-454f-9059-0b6f7e792216>
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/ekttrank-0000006f", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("PJSIP/ekttrank-0000006f", "PJSIP/180-00000070 montior file= ") in new stack
-- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/ekttrank-0000006f", "1?skipagi") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("PJSIP/ekttrank-0000006f", "") in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/ekttrank-0000006f' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/ekttrank-0000006f'
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>
opnPBX2*CLI>

ded · Сообщение **ded** » 27 май 2021, 12:56

freeman33 писал(а):я понимаю что вы тут все специалисты и диагностировали бы проблему за 5 мин.,

Нет.

freeman33 писал(а): но я затем сюда и обратился чтоб разъяснили в чем дело.

Это вы попутали, уважаемый, тут не платный суппорт.
Вы кто, Роман Викторович Шум? Создаёте шум? повышенное внимание к своей персоне? Индивидуальное обучение через форум специалистов?

Картина мутная. трассировку показываете по внутренним ИП адресам

traceroute to 172.18.253.1 (172.18.253.1), 30 hops max, 60 byte packets
1 gateway (192.168.103.253) 0.076 ms 0.043 ms 0.030 ms
2 172.20.20.101 (172.20.20.101) 23.930 ms 23.924 ms 24.131 ms
3 172.18.253.1 (172.18.253.1) 25.006 ms 24.948 ms 25.047 ms

А в поле контакт во многих местах что-то зашхериваете крестиками, там что, ещё и публичный ИП адрес?
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@xx.xxx.xxx.xx:5060

В Астериске есть внутренний механизм определения - куда посылать, в мир, через публичный адрес externip=, или без НАТа, по сетям из параметра localnet=
https://www.voip-info.org/asterisk-sip-externip/
возможно ваши сети нужно туда просто прописать, 192.168.0.0/16 и 172.16.0.0/12

Напомню то, с чем вы согласились при регистрации на форуме:

Если Вы регистрируетесь тут, чтобы первым делом написать "Я в Астериске - нуб", или "Я новичок, уже третий день бьюсь, не пинайте сильно!" то воздержитесь от регистрации либо размещайте Ваше сообщение только в разделе "Бизнес".

Новичком тут считается только прочитавший (как минимум) «Астериск - будущее телефонии» и пытающийся сделать большее.
Отсутствие необходимого минимума знаний НЕ является Вашим оправданием.

Скорее всего Вам просто нужна помощь, но помощь уже перед Вами - Вы можете (и должны) искать похожие случаи, а они обязательно есть, и как они были преодолены.

freeman33 · Сообщение **freeman33** » 27 май 2021, 13:17

ded писал(а):
freeman33 писал(а):я понимаю что вы тут все специалисты и диагностировали бы проблему за 5 мин.,
Нет.
freeman33 писал(а): но я затем сюда и обратился чтоб разъяснили в чем дело.
Это вы попутали, уважаемый, тут не платный суппорт.

Картина мутная. трасировку показываете по внутренним ИП адресам
traceroute to 172.18.253.1 (172.18.253.1), 30 hops max, 60 byte packets
1 gateway (192.168.103.253) 0.076 ms 0.043 ms 0.030 ms
2 172.20.20.101 (172.20.20.101) 23.930 ms 23.924 ms 24.131 ms
3 172.18.253.1 (172.18.253.1) 25.006 ms 24.948 ms 25.047 ms
А в поле контакт во многих местах что-то зашхериваете крестиками, там что, ещё и публичный ИП адрес?
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@xx.xxx.xxx.xx:5060

В Астериске есть внутренний механизм определения - куда посылать, в мир, через публичный адрес externip=, или без НАТа, по сетям из параметра localnet=
https://www.voip-info.org/asterisk-sip-externip/
возможно ваши сети нужно туда просто прописать, 192.168.0.0/16 и 172.16.0.0/12

Напомню то, с чем вы согласились при регистрации на форуме:
Если Вы регистрируетесь тут, чтобы первым делом написать "Я в Астериске - нуб", или "Я новичок, уже третий день бьюсь, не пинайте сильно!" то воздержитесь от регистрации либо размещайте Ваше сообщение только в разделе "Бизнес".

Новичком тут считается только прочитавший (как минимум) «Астериск - будущее телефонии» и пытающийся сделать большее.
Отсутствие необходимого минимума знаний НЕ является Вашим оправданием.

Скорее всего Вам просто нужна помощь, но помощь уже перед Вами - Вы можете (и должны) искать похожие случаи, а они обязательно есть, и как они были преодолены.

Да, согласился )
Между филиалом и центром IPSec на микротиках.
Да, есть публичный. Он меня тоже смущает, в туннеле должны только внутренние адреса ходить, а это получается адрес пира, с которым строится туннель, на стороне филиала.

sip.conf посмотрел, пусто там, все закоментарено кроме [general].
А может надо pjsip.conf смотреть, т.к. у нас эта библиотека используется?

Книжки люблю читать, просто сейчас нет времени перелопатить ее, так же как и тестовый астер поднять.

Zavr2008 · Сообщение **Zavr2008** » 27 май 2021, 13:42

Оверквоттинг показывает полное неуважение данного товарища к остальным..

А в поле контакт во многих местах что-то зашхериваете крестиками, там что, ещё и публичный ИП адрес?

Как и подобное. Серые адреса замалевывать - это та еще тема..
Если белые там адреса свои - значит не верно настроен local_net и extern ip.

В портянке лога консоли видно сколько раз безуспешно посылается INVITE и вторая сторона его походу не получает.
В результате и отправляется отлуп.

Какой момент должен писать Asterisk про Critical packet retransmission ?

Это в chan_sip, я к сожалению не досмотрел что ТС повелся на новомодное)

freeman33 · Сообщение **freeman33** » 27 май 2021, 13:57

freeman33 писал(а): но я затем сюда и обратился чтоб разъяснили в чем дело.

Это вы попутали, уважаемый, тут не платный суппорт.
Вы кто, Роман Викторович Шум? Создаёте шум? повышенное внимание к своей персоне? Индивидуальное обучение через форум специалистов?
[/size][/quote][/quote]

Причем тут вышеобозначенное ФИО, вы на личности хотите перейти что ли? Так вот это, не он, к вашему сведению.

Zavr2008 · Сообщение **Zavr2008** » 28 май 2021, 12:44

ТС, все тут ежедневно видим закипающие чайники.
Лучше начните изучать тему, книжки читать и старайтесь поменьше нарушать правила.
Тогда вопросы будут конкретные, ну и ответы более точные.

Ну или есть другой путь если "некогда вникать" - платный саппорт. Ваша проблема не такая и сложная для тех кто знает что делает.

freeman33 · Сообщение **freeman33** » 28 май 2021, 13:32

Все нормально )

Zavr2008 · Сообщение **Zavr2008** » 28 май 2021, 14:33

Всё нормально))

Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек

Re: Пропадает связь через 30 сек