Проблема Retransmission в одной подсети Elastix

RUMarat · Сообщение **RUMarat** » 12 ноя 2015, 22:07

Ребята! Коллеги! Друзья!
Выручайте.
Есть простая схема
ТфОП <---> E1 <---> Шлюз E1 to SIP <---> Elastix <---> IP Phone <2800>
//___________ (192.168.100.2) <---> (192.168.100.45) <---> (192.168.100.58)

Поступает входящий звонок на городской номер 68-57-47, терминируем его на экстеншн 2800 – все ОК
Пробуем навести его сначала на приветствие, после чего на 2800 – общая продолжительность звонка 32 секунды и сброс, при этом в логах появляется
WARNING[4189]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 76ec-f8e7-fef4-2226@192.168.100.2 for seqno 101 (Critical Response)
Packet timed out after 31999ms with no response

Настройка транка

Код: Выделить всё

host=192.168.100.2
port=5070
nat=never
type=friend
canreinvite=no
dtfmmode=inband

Откуда проблема с Retransmission если все устройства в одной сети?

UPDATE:
Пытался перенастроить входящий маршрут и эксперимента ради ткнул галочку на «Signal RINGING», не помогло, а вместо приветствия пошли только гудки, ототкнул обратно галочку а всё равно только гудки

Возник второй вопрос: что поменяла эта галочка и как это вернуть как было?

Спасибо!

SIP Debug прилагаю, но уже после галочки. До этого он был немного другой, но чёт не найду предыдущий

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: SIP Debug после этой галочки:

elastix*CLI> sip set debug on
SIP Debugging enabled
-- Remote UNIX connection
-- Remote UNIX connection disconnected

<--- SIP read from UDP:192.168.100.2:5070 --->
INVITE sip:685747@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK6A7F6321786733344
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>
Contact: <sip:961хххх597@192.168.100.2>
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Expires: 180
Supported: replaces
Accept: application/sdp
Content-Length: 207
Content-Type: application/sdp
CT-ExtLevel: 1
Record-Route: <sip:192.168.100.2:5070;lr>

v=0
o=CTBFv1.0 20788 0 IN IP4 192.168.100.2
s=SIP Call
c=IN IP4 192.168.100.2
t=0 0
m=audio 8334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:30
<------------->
--- (16 headers 11 lines) ---
Sending to 192.168.100.2:5070 (no NAT)
Sending to 192.168.100.2:5070 (no NAT)
Using INVITE request as basis request - b285-3d12-7a2b-a1ed@192.168.100.2
Found peer 'SiTi_IP_M' for '961хххх597' from 192.168.100.2:5070
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|slin|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.2:8334
Looking for 685747 in from-internal (domain elastix.56.to.fskn)
list_route: hop: <sip:192.168.100.2:5070;lr>

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:685747@192.168.100.45:5060>
Content-Length: 0

<------------>
-- Executing [685747@from-internal:1] ResetCDR("SIP/SiTi_IP_M-0000083d", "") in new stack
-- Executing [685747@from-internal:2] NoCDR("SIP/SiTi_IP_M-0000083d", "") in new stack
-- Executing [685747@from-internal:3] Progress("SIP/SiTi_IP_M-0000083d", "") in new stack
Audio is at 18854
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:685747@192.168.100.45:5060>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 183384943 183384943 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 18854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Executing [685747@from-internal:4] Wait("SIP/SiTi_IP_M-0000083d", "1") in new stack
> 0x2b4b7c111240 -- Probation passed - setting RTP source address to 192.168.100.2:8334
-- Executing [685747@from-internal:5] Progress("SIP/SiTi_IP_M-0000083d", "") in new stack
-- Executing [685747@from-internal:6] Playback("SIP/SiTi_IP_M-0000083d", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/SiTi_IP_M-0000083d> Playing 'silence/1.slin' (language 'ru')
-- <SIP/SiTi_IP_M-0000083d> Playing 'cannot-complete-as-dialed.slin' (language 'ru')
-- <SIP/SiTi_IP_M-0000083d> Playing 'check-number-dial-again.slin' (language 'ru')
-- Executing [685747@from-internal:7] Wait("SIP/SiTi_IP_M-0000083d", "1") in new stack
-- Executing [685747@from-internal:8] Congestion("SIP/SiTi_IP_M-0000083d", "20") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2015-11-12 23:45:15] WARNING[20368][C-0000080f]: channel.c:4860 ast_prod: Prodding channel 'SIP/SiTi_IP_M-0000083d' failed
== Spawn extension (from-internal, 685747, 8) exited non-zero on 'SIP/SiTi_IP_M-0000083d'
-- Executing [h@from-internal:1] Hangup("SIP/SiTi_IP_M-0000083d", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/SiTi_IP_M-0000083d'
Retransmitting #1 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #2 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #3 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #4 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #5 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (no NAT) to 13.0.0.127

OPTIONS sip:15622801@13.0.0.127:39269;rinstance=df90d785de7a764c;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK767d8c1e
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as7d42a372
To: <sip:15622801@13.0.0.127:39269;rinstance=df90d785de7a764c;transport=UDP>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 4be11d9e16fc863144b0739d3f037050@192.168.100.45:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 12 Nov 2015 18:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:13.0.0.127:39269 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK767d8c1e
Contact: <sip:13.0.0.127:39269>
To: <sip:15622801@13.0.0.127:39269;rinstance=df90d785de7a764c;transport=UDP>;tag=0a07890a
From: "Unknown"<sip:Unknown@192.168.100.45>;tag=as7d42a372
Call-ID: 4be11d9e16fc863144b0739d3f037050@192.168.100.45:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '4be11d9e16fc863144b0739d3f037050@192.168.100.45:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
OPTIONS sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK50dd779e
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as23499de5
To: <sip:2800@192.168.100.58:5060>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 5bcfee0d067c80e039e27acf18f5dbb2@192.168.100.45:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 12 Nov 2015 18:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=d891bae1925f5c40i0
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as23499de5
Call-ID: 5bcfee0d067c80e039e27acf18f5dbb2@192.168.100.45:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK50dd779e
Server: Cisco/SPA504G-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5bcfee0d067c80e039e27acf18f5dbb2@192.168.100.45:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 10.6.80.6:5060:
OPTIONS sip:10.6.80.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK45e66222
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as0ed60c05
To: <sip:10.6.80.6>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 5a443a8f74dc1f1c2414840b1d8adb67@192.168.100.45:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 12 Nov 2015 18:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:10.6.80.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK45e66222
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as0ed60c05
Call-ID: 5a443a8f74dc1f1c2414840b1d8adb67@192.168.100.45:5060
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.0.0 build 829
To: <sip:10.6.80.6>;tag=aab38cdd1066946a
Content-Type: application/sdp
Content-Length: 239

v=0
o=UserA 3990161992 2497974283 IN IP4 13.0.0.147
s=Session SDP
c=IN IP4 13.1.0.147
t=0 0
m=audio 8000 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/16000
<------------->
--- (11 headers 11 lines) ---
Really destroying SIP dialog '5a443a8f74dc1f1c2414840b1d8adb67@192.168.100.45:5060' Method: OPTIONS
Retransmitting #6 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #7 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #8 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #9 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #10 (no NAT) to 192.168.100.2:5070:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK6A7F6321786733344;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK46CC915D1825117648;received=192.168.100.2
From: <sip:961хххх597@Orenburg>;tag=11878111227464
To: <sip:685747@Orenburg>;tag=as22480229
Call-ID: b285-3d12-7a2b-a1ed@192.168.100.2
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---
[2015-11-12 23:45:47] WARNING[4189]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission b285-3d12-7a2b-a1ed@192.168.100.2 for seqno 101 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 31999ms with no response
Really destroying SIP dialog 'b285-3d12-7a2b-a1ed@192.168.100.2' Method: INVITE

ded · Сообщение **ded** » 13 ноя 2015, 00:12

Всё плохо.
Всё через Ж, (и это ж.... - неспроста!)
С одной стороны - создали себе дополнительный гимор на шлюзе,сообщив ему, что он будет работать через сигнальный порт 5070.И он как бы не против, но при этом знает и помнит о родном 5060 -
Via: SIP/2.0/UDP 192.168.100.2:5060 - сообщает об этом!

Но это не главное. выполнение диалплана 685747@from-internal на Астериске затыкается на шагах 4-5-6, и это состояние Астериск пытается отправить шлюзу Е1 (на порт 5070),но он слушать уже не хочет,видать закрыл диалог.

Код: Выделить всё

- Executing [685747@from-internal:4] Wait("SIP/SiTi_IP_M-0000083d", "1") in new stack
> 0x2b4b7c111240 -- Probation passed - setting RTP source address to 192.168.100.2:8334
-- Executing [685747@from-internal:5] Progress("SIP/SiTi_IP_M-0000083d", "") in new stack
-- Executing [685747@from-internal:6] Playback("SIP/SiTi_IP_M-0000083d", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/SiTi_IP_M-0000083d> Playing 'silence/1.slin' (language 'ru')
-- <SIP/SiTi_IP_M-0000083d> Playing 'cannot-complete-as-dialed.slin' (language 'ru')
-- <SIP/SiTi_IP_M-0000083d> Playing 'check-number-dial-again.slin' (language 'ru')

Скорее всего у вас перепиленый исковерканный кем-то freePBX, изучать его покуроченный диалплан - тускло

Ну и до кучи - что это там за ИП адресация присутствует?
<--- SIP read from UDP:13.0.0.127:39269 --->

Zavr2008 · Сообщение **Zavr2008** » 13 ноя 2015, 01:41

RUMarat писал(а): Есть простая схема
ТфОП <---> E1 <---> Шлюз E1 to SIP <---> Elastix <---> IP Phone <2800>
//___________ (192.168.100.2) <---> (192.168.100.45) <---> (192.168.100.58)

Это - кривая схема. Объясняю:

User-Agent: CTBFv1.0

Если я не ошибаюсь, под "Шлюз Е1 to SIP" дружно подкралась Avaya IPO500. Я прав?

RUMarat · Сообщение **RUMarat** » 13 ноя 2015, 07:20

ded писал(а):С одной стороны - создали себе дополнительный гимор на шлюзе,сообщив ему, что он будет работать через сигнальный порт 5070.И он как бы не против, но при этом знает и помнит о родном 5060 -
Via: SIP/2.0/UDP 192.168.100.2:5060 - сообщает об этом!

этот "гимор" был предустановлен на этом шлюзе еще производителем, и с другими устройствами он проработал лет 8 уже, правда без IVRов и Приветствий

Код: Выделить всё

Настройки SIP-сервера (выдержка):
Номер собственного порта  5060
Номер порта прокси-сервера 5070
Адрес прокси-сервера 192.168.100.2

и здесь вполне возможно надо на транке * поставить порт 5060, согласен.

ded писал(а):выполнение диалплана 685747@from-internal на Астериске затыкается на шагах 4-5-6, и это состояние Астериск пытается отправить шлюзу Е1 (на порт 5070),но он слушать уже не хочет,видать закрыл диалог

Эта ситуация возникла после той злополучной галочки, поэтому и не трогаю сейчас настройки прохождения входящих вызовов до того, как не решу (решим

) проблему с отсутствием "голоса" автоинформатора и наличием вместо онного сигнала КПВ.

ded писал(а):Скорее всего у вас перепиленый исковерканный кем-то freePBX, изучать его покуроченный диалплан - тускло
Ну и до кучи - что это там за ИП адресация присутствует?
<--- SIP read from UDP:13.0.0.127:39269 --->

Нет, не особо исковерканный (мной по крайней мере

). Ставил пакет Elastix в сборе. Добавил только "Манипуляции с CID при транзите вызова" и не более того. А 13.0.0.127 это у нас подсеть VPN, но она у нас с локальной сетью работает без NAT преспокойненько и * об этом знает.

Код: Выделить всё

Выдержка из SIP conf:
[general]
localnet=13.0.0.0/8
localnet=192.0.0.0/8

Там на машине софтфон стоит, он и промелькнул в дебаге, но в данной ситуации его это не касается

RUMarat · Сообщение **RUMarat** » 13 ноя 2015, 07:29

Zavr2008 писал(а):Если я не ошибаюсь, под "Шлюз Е1 to SIP" дружно подкралась Avaya IPO500. Я прав?

А вот и ошибаетесь

Avaya IPO500 конечно же там присутствует, она стоит в удаленном подразделении, но в нашей сети (192.168.98.6), именно с её CID мы и манипулирвали

Но в данной ситуации в роли "Шлюз Е1 to SIP" выступает CиТи-IP-M

virus_net · Сообщение **virus_net** » 13 ноя 2015, 07:50

RUMarat писал(а):localnet=13.0.0.0/8
localnet=192.0.0.0/8

Сильно !

RUMarat · Сообщение **RUMarat** » 13 ноя 2015, 08:04

Бесспорно, но VPN у нас настроен централизованно и с эти уже ничего не поделаешь

, а так как всё локальной сети то и не страшно

RUMarat · Сообщение **RUMarat** » 13 ноя 2015, 09:13

Помогите хотя бы с этим: что поменяла эта галочка (Signal RINGING) и как это вернуть как было?
А потом можно будет разобраться с остальным

ded · Сообщение **ded** » 13 ноя 2015, 11:43

Сделайте не обратно как было, а правильно.
В схеме
ТфОП <---> E1 <---> Шлюз E1 to SIP <---> Elastix <---> IP Phone <2800>
шлюз должен втыкаться как транк с внешней стороны Эластика, то есть должен иметь контекст from-pstn, тогда уже настроить входящую маршрутизацию, IVR, и прочие штуки. А Вы его воткнули сразу в абонентскую часть - from-internal. Это работать как надо не будет.

На будущее: если освоили копипасту - логи сюда паблишить тонну, имейте совесть: убирайте лишнее, чтобы не ломать голову - что такое за пакеты из 13.0.0.127

RUMarat · Сообщение **RUMarat** » 13 ноя 2015, 13:07

Прошу прощения за тонны логов, да ещё и с мусором. Для полноты картины оставлял всё.

Пробовал ставить контекст=[from-pstn] и [from-trunk] на входящем плече транка, ситуация не менялась

Попробовал в контексте [from-internal-custom] прописать команды по аналогии с создаваемым автоматически FreePBX'ом:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: from-internal-custom

Код: Выделить всё

exten => 685747,1,Set(__FROM_DID=${EXTEN})
exten => 685747,n,Set(CHANNEL(language)=RU)
exten => 685747,n,Gosub(app-blacklist-check,s,1())
exten => 685747,n,Set(__REC_POLICY_MODE=always)
exten => 685747,n,Set(CDR(did)=${FROM_DID})
exten => 685747,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 685747,n,Set(CHANNEL(musicclass)=default)
exten => 685747,n,Set(__MOHCLASS=default)
exten => 685747,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 685747,n,Set(CALLERPRES()=allowed_not_screened)
exten => 685747,n,Answer
exten => 685747,n,Wait(1)
exten => 685747,n,Noop(Playing announcement 3445)
exten => 685747,n,Playback(custom/tel_dov,noanswer)
exten => 685747,n,Goto(from-did-direct,2800,1)

в итоге голос пошел, но всё тот же обрыв на 32 секунде.

Лог с сип-дебагом прилагаю:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Снова тонна, но без мусора

elastix*CLI> sip set debug peer SiTi_IP_M_in
SIP Debugging Enabled for IP: 192.168.100.2
elastix*CLI> sip set debug peer 2800
SIP Debugging Enabled for IP: 192.168.100.58
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [685747@from-internal:1] Set("SIP/SiTi_IP_M-00000a9f", "__FROM_DID=685747") in new stack
-- Executing [685747@from-internal:2] Set("SIP/SiTi_IP_M-00000a9f", "CHANNEL(language)=RU") in new stack
-- Executing [685747@from-internal:3] Gosub("SIP/SiTi_IP_M-00000a9f", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/SiTi_IP_M-00000a9f", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [685747@from-internal:4] Set("SIP/SiTi_IP_M-00000a9f", "__REC_POLICY_MODE=always") in new stack
-- Executing [685747@from-internal:5] Set("SIP/SiTi_IP_M-00000a9f", "CDR(did)=685747") in new stack
-- Executing [685747@from-internal:6] ExecIf("SIP/SiTi_IP_M-00000a9f", "1 ?Set(CALLERID(name)=3532685720)") in new stack
-- Executing [685747@from-internal:7] Set("SIP/SiTi_IP_M-00000a9f", "CHANNEL(musicclass)=default") in new stack
-- Executing [685747@from-internal:8] Set("SIP/SiTi_IP_M-00000a9f", "__MOHCLASS=default") in new stack
-- Executing [685747@from-internal:9] Set("SIP/SiTi_IP_M-00000a9f", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [685747@from-internal:10] Set("SIP/SiTi_IP_M-00000a9f", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [685747@from-internal:11] Answer("SIP/SiTi_IP_M-00000a9f", "") in new stack
> 0x2b4b5c07b5c0 -- Probation passed - setting RTP source address to 192.168.100.2:8336
-- Executing [685747@from-internal:12] Wait("SIP/SiTi_IP_M-00000a9f", "1") in new stack
-- Executing [685747@from-internal:13] NoOp("SIP/SiTi_IP_M-00000a9f", "Playing announcement 3445") in new stack
-- Executing [685747@from-internal:14] Playback("SIP/SiTi_IP_M-00000a9f", "custom/tel_dov,noanswer") in new stack
-- <SIP/SiTi_IP_M-00000a9f> Playing 'custom/tel_dov.slin' (language 'RU')
-- Executing [685747@from-internal:15] Goto("SIP/SiTi_IP_M-00000a9f", "from-did-direct,2800,1") in new stack
-- Goto (from-did-direct,2800,1)
-- Executing [2800@from-did-direct:1] Set("SIP/SiTi_IP_M-00000a9f", "__RINGTIMER=15") in new stack
-- Executing [2800@from-did-direct:2] Macro("SIP/SiTi_IP_M-00000a9f", "exten-vm,novm,2800,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/SiTi_IP_M-00000a9f", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/SiTi_IP_M-00000a9f", "TOUCH_MONITOR=1447408255.7615") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/SiTi_IP_M-00000a9f", "AMPUSER=3532685720") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/SiTi_IP_M-00000a9f", "1?Set(REALCALLERIDNUM=3532685720)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/SiTi_IP_M-00000a9f", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/SiTi_IP_M-00000a9f", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/SiTi_IP_M-00000a9f", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/SiTi_IP_M-00000a9f", "CALLERID(number)=3532685720") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/SiTi_IP_M-00000a9f", "CALLERID(name)=3532685720") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/SiTi_IP_M-00000a9f", "CDR(cnum)=3532685720") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/SiTi_IP_M-00000a9f", "CDR(cnam)=3532685720") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/SiTi_IP_M-00000a9f", "CHANNEL(language)=RU") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/SiTi_IP_M-00000a9f", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/SiTi_IP_M-00000a9f", "__EXTTOCALL=2800") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/SiTi_IP_M-00000a9f", "__PICKUPMARK=2800") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/SiTi_IP_M-00000a9f", "RT=") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/SiTi_IP_M-00000a9f", "sub-record-check,s,1(exten,2800,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/SiTi_IP_M-00000a9f", "REC_POLICY_MODE_SAVE=always") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/SiTi_IP_M-00000a9f", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/SiTi_IP_M-00000a9f", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/SiTi_IP_M-00000a9f", "NOW=1447408264") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/SiTi_IP_M-00000a9f", "__DAY=13") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/SiTi_IP_M-00000a9f", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/SiTi_IP_M-00000a9f", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/SiTi_IP_M-00000a9f", "__TIMESTR=20151113-145104") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/SiTi_IP_M-00000a9f", "__FROMEXTEN=3532685720") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/SiTi_IP_M-00000a9f", "__CALLFILENAME=exten-2800-3532685720-20151113-145104-1447408255.7615") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/SiTi_IP_M-00000a9f", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?callee") in new stack
-- Goto (sub-record-check,exten,8)
-- Executing [exten@sub-record-check:8] GosubIf("SIP/SiTi_IP_M-00000a9f", "1?record,1(exten,2800,3532685720)") in new stack
-- Executing [record@sub-record-check:1] Set("SIP/SiTi_IP_M-00000a9f", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [record@sub-record-check:2] MixMonitor("SIP/SiTi_IP_M-00000a9f", "2015/11/13/exten-2800-3532685720-20151113-145104-1447408255.7615.wav,,") in new stack
-- Executing [record@sub-record-check:3] Set("SIP/SiTi_IP_M-00000a9f", "__REC_STATUS=RECORDING") in new stack
-- Executing [record@sub-record-check:4] Set("SIP/SiTi_IP_M-00000a9f", "CDR(recordingfile)=exten-2800-3532685720-20151113-145104-1447408255.7615.wav") in new stack
-- Executing [record@sub-record-check:5] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [exten@sub-record-check:9] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [s@macro-exten-vm:7] Macro("SIP/SiTi_IP_M-00000a9f", "dial-one,,Ttr,2800") in new stack
== Begin MixMonitor Recording SIP/SiTi_IP_M-00000a9f
-- Executing [s@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000a9f", "DEXTEN=2800") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000a9f", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/SiTi_IP_M-00000a9f", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/SiTi_IP_M-00000a9f", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000a9f", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/SiTi_IP_M-00000a9f", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000a9f", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000a9f", "DEVICES=2800") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Set(DEVICES=5622800)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000a9f", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/SiTi_IP_M-00000a9f", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000a9f", "THISDIAL=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/SiTi_IP_M-00000a9f", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000a9f", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/SiTi_IP_M-00000a9f", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/SiTi_IP_M-00000a9f", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000a9f", "THISPART2=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?Set(THISPART2=DAHDI/2800)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000a9f", "NEWDIAL=SIP/2800&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/SiTi_IP_M-00000a9f", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000a9f", "THISDIAL=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/SiTi_IP_M-00000a9f", "DSTRING=SIP/2800&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000a9f", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/SiTi_IP_M-00000a9f", "DSTRING=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/SiTi_IP_M-00000a9f", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000a9f", "DB(CALLTRACE/2800)=3532685720") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/SiTi_IP_M-00000a9f", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/SiTi_IP_M-00000a9f", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/SiTi_IP_M-00000a9f", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/SiTi_IP_M-00000a9f", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/SiTi_IP_M-00000a9f", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/SiTi_IP_M-00000a9f", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/SiTi_IP_M-00000a9f", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/SiTi_IP_M-00000a9f", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:38] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?godial") in new stack
-- Goto (macro-dial-one,s,43)
-- Executing [s@macro-dial-one:43] Dial("SIP/SiTi_IP_M-00000a9f", "SIP/2800,,Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 13352
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
INVITE sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK3c25585e
Max-Forwards: 70
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
To: <sip:2800@192.168.100.58:5060>
Contact: <sip:3532685720@192.168.100.45:5060>
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Fri, 13 Nov 2015 09:51:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 1252547186 1252547186 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 13352 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/2800

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 100 Trying
To: <sip:2800@192.168.100.58:5060>
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK3c25585e
Server: Cisco/SPA504G-7.5.2
Content-Length: 0

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

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 180 Ringing
To: <sip:2800@192.168.100.58:5060>;tag=2aae9d9eba5eccfai0
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK3c25585e
Contact: "2800" <sip:2800@192.168.100.58:5060>
Server: Cisco/SPA504G-7.5.2
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:2800@192.168.100.58:5060>
-- SIP/2800-00000aa0 is ringing
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
OPTIONS sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK1bc43142
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as5183f0e6
To: <sip:2800@192.168.100.58:5060>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 75a7c124196b70cd1cdf17ea4885fdc9@192.168.100.45:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Fri, 13 Nov 2015 09:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=d891bae1925f5c40i0
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as5183f0e6
Call-ID: 75a7c124196b70cd1cdf17ea4885fdc9@192.168.100.45:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK1bc43142
Server: Cisco/SPA504G-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '75a7c124196b70cd1cdf17ea4885fdc9@192.168.100.45:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=2aae9d9eba5eccfai0
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK3c25585e
Contact: "2800" <sip:2800@192.168.100.58:5060>
Server: Cisco/SPA504G-7.5.2
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 26725249 26725249 IN IP4 192.168.100.58
s=-
c=IN IP4 192.168.100.58
t=0 0
m=audio 16470 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.58:16470
list_route: hop: <sip:2800@192.168.100.58:5060>
set_destination: Parsing <sip:2800@192.168.100.58:5060> for address/port to send to
set_destination: set destination to 192.168.100.58:5060
Transmitting (no NAT) to 192.168.100.58:5060:
ACK sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK50b5a310
Max-Forwards: 70
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
To: <sip:2800@192.168.100.58:5060>;tag=2aae9d9eba5eccfai0
Contact: <sip:3532685720@192.168.100.45:5060>
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0

---
-- SIP/2800-00000aa0 answered SIP/SiTi_IP_M-00000a9f
> 0x2b4b780ac210 -- Probation passed - setting RTP source address to 192.168.100.58:16470

<--- SIP read from UDP:192.168.100.58:5060 --->
REGISTER sip:192.168.100.45:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.58:5060;branch=z9hG4bK-4bdb843
From: "2800" <sip:2800@192.168.100.45>;tag=a233926dba68a24o0
To: "2800" <sip:2800@192.168.100.45>
Call-ID: 35d61fa-b9f1af04@192.168.100.58
CSeq: 46661 REGISTER
Max-Forwards: 70
Authorization: Digest username="2800",realm="asterisk",nonce="3bf8d176",uri="sip:192.168.100.45:5060",algorithm=MD5,response="991ef28723e67ebcea68c795c0f9ec35"
Contact: "2800" <sip:2800@192.168.100.58:5060>;expires=3600
User-Agent: Cisco/SPA504G-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.58:5060 (no NAT)
Sending to 192.168.100.58:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.58:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.58:5060;branch=z9hG4bK-4bdb843;received=192.168.100.58
From: "2800" <sip:2800@192.168.100.45>;tag=a233926dba68a24o0
To: "2800" <sip:2800@192.168.100.45>;tag=as63b44f7c
Call-ID: 35d61fa-b9f1af04@192.168.100.58
CSeq: 46661 REGISTER
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0564afb2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '35d61fa-b9f1af04@192.168.100.58' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.58:5060 --->
REGISTER sip:192.168.100.45:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.58:5060;branch=z9hG4bK-4e00f444
From: "2800" <sip:2800@192.168.100.45>;tag=a233926dba68a24o0
To: "2800" <sip:2800@192.168.100.45>
Call-ID: 35d61fa-b9f1af04@192.168.100.58
CSeq: 46662 REGISTER
Max-Forwards: 70
Authorization: Digest username="2800",realm="asterisk",nonce="0564afb2",uri="sip:192.168.100.45:5060",algorithm=MD5,response="bc1eb91201c97bbf41791f3edc328e31"
Contact: "2800" <sip:2800@192.168.100.58:5060>;expires=3600
User-Agent: Cisco/SPA504G-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.58:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
OPTIONS sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK5c3b398b
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as02bbb6dd
To: <sip:2800@192.168.100.58:5060>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 1781ee5b30ac22c65386b4b47dba6e04@192.168.100.45:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Fri, 13 Nov 2015 09:51:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (no NAT) to 192.168.100.58:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.58:5060;branch=z9hG4bK-4e00f444;received=192.168.100.58
From: "2800" <sip:2800@192.168.100.45>;tag=a233926dba68a24o0
To: "2800" <sip:2800@192.168.100.45>;tag=as63b44f7c
Call-ID: 35d61fa-b9f1af04@192.168.100.58
CSeq: 46662 REGISTER
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:2800@192.168.100.58:5060>;expires=3600
Date: Fri, 13 Nov 2015 09:51:26 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '276c09857dac95167f9afd0719cf5cb4@192.168.100.45:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
NOTIFY sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK5a1e49f0
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as1d3ebf39
To: <sip:2800@192.168.100.58:5060>
Contact: <sip:Unknown@192.168.100.45:5060>
Call-ID: 276c09857dac95167f9afd0719cf5cb4@192.168.100.45:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.11.0(11.13.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 85

Messages-Waiting: no
Message-Account: sip:*97@192.168.100.45
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '35d61fa-b9f1af04@192.168.100.58' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=d891bae1925f5c40i0
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as02bbb6dd
Call-ID: 1781ee5b30ac22c65386b4b47dba6e04@192.168.100.45:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK5c3b398b
Server: Cisco/SPA504G-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1781ee5b30ac22c65386b4b47dba6e04@192.168.100.45:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=d891bae1925f5c40i0
From: "Unknown" <sip:Unknown@192.168.100.45>;tag=as1d3ebf39
Call-ID: 276c09857dac95167f9afd0719cf5cb4@192.168.100.45:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK5a1e49f0
Server: Cisco/SPA504G-7.5.2
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '276c09857dac95167f9afd0719cf5cb4@192.168.100.45:5060' Method: NOTIFY
[2015-11-13 14:51:27] WARNING[4189]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 1849-9a23-22d1-1680@192.168.100.2 for seqno 101 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[2015-11-13 14:51:27] WARNING[4189]: chan_sip.c:4053 retrans_pkt: Hanging up call 1849-9a23-22d1-1680@192.168.100.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
-- Executing [h@macro-dial-one:1] Macro("SIP/SiTi_IP_M-00000a9f", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/SiTi_IP_M-00000a9f", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/SiTi_IP_M-00000a9f", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/SiTi_IP_M-00000a9f", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/SiTi_IP_M-00000a9f", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/SiTi_IP_M-00000a9f", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/SiTi_IP_M-00000a9f", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/SiTi_IP_M-00000a9f>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/SiTi_IP_M-00000a9f", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/SiTi_IP_M-00000a9f' in macro 'hangupcall'
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/SiTi_IP_M-00000a9f'
Scheduling destruction of SIP dialog '6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2800@192.168.100.58:5060> for address/port to send to
set_destination: set destination to 192.168.100.58:5060
Reliably Transmitting (no NAT) to 192.168.100.58:5060:
BYE sip:2800@192.168.100.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK4126209e
Max-Forwards: 70
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
To: <sip:2800@192.168.100.58:5060>;tag=2aae9d9eba5eccfai0
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.13.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/SiTi_IP_M-00000a9f' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/SiTi_IP_M-00000a9f' in macro 'exten-vm'
== Spawn extension (from-did-direct, 2800, 2) exited non-zero on 'SIP/SiTi_IP_M-00000a9f'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/SiTi_IP_M-00000a9f

<--- SIP read from UDP:192.168.100.58:5060 --->
SIP/2.0 200 OK
To: <sip:2800@192.168.100.58:5060>;tag=2aae9d9eba5eccfai0
From: "3532685720" <sip:3532685720@192.168.100.45>;tag=as135db13e
Call-ID: 6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK4126209e
Server: Cisco/SPA504G-7.5.2
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6e5d42b45e1549f20d8bf822100612fb@192.168.100.45:5060' Method: INVITE
elastix*CLI> sip set debug off
SIP Debugging Disabled

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