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Проблема Retransmission в одной подсети Elastix

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Модераторы: april22, Zavr2008

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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: Проблема Retransmission в одной подсети Elastix

Сообщение ded »

Пробовал ставить контекст=[from-pstn] и [from-trunk] на входящем плече транка, ситуация не менялась
не информативно.
Ставьте контекст=[from-pstn] и делайте вызов. Смотрите при этом в консоль (или потом в лог).
Пока что по-прежнему во внутреннем контексте
- Executing [685747@from-internal

SIP дебаг приведён между 192.168.100.58 и 192.168.100.45, а затык (retransmissions) с адресом шлюза 192.168.100.2
Нет ли у вас там qualify=yes на пире шлюза? Если он ВДРУГ не отвечает на пакеты OPTIONS - это может быть проблемой.
Уберите вообще пакеты OPTIONS. Они в локальной сети ни к чему.
qualify=no
Вернуться к началу
RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

ded писал(а):между 192.168.100.58 и 192.168.100.45, а затык (retransmissions) с адресом шлюза 192.168.100.2
Вот она, пятница тринадцатое :)

Хотел дебаг сразу с двумя пирами сделать.

Попробуем ещё контексты поменять, посравнивать реакции в логе.

А from-pstn он же included в from-internal вроде.
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: Проблема Retransmission в одной подсети Elastix

Сообщение ded »

RUMarat писал(а):А from-pstn он же included в from-internal вроде.
Вот оно, уже на пятницу тринадцатое не свалить! :)
Тест на администрирование freePBX не пройден.
Вернуться к началу
RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

extensions.conf гласит:

Код: Выделить всё

[from-internal]
include => from-internal-noxfer
include => from-internal-xfer
include => bad-number ; auto-generated
include => from-trunk 
 ...

[from-trunk]
include => from-pstn
то есть в итоге он инклюдится, или я неверно рассуждаю?

off:
Книгу читал неоднократно, просто одновременно приходится вести hicom, avaya, СиТи, *, администрировать сеть, прокладывать видеонаблюдение, ремонтировать сигнализацию, это еще не сказано про бумажную работу :) поэтому могу подпутывать порой :)
Вернуться к началу
RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

Настройка транка в вебинтерфейсе:

Incoming Settings

USER Context?: SiTi_IP_M_in

Код: Выделить всё

host=192.168.100.2
port=5070
nat=never
type=friend
canreinvite=no
dtfmmode=inband
relaxdtmf=yes
context=from-pstn
настроен Inbound Route для номера 685747:
приветствие -> внутренний номер 2800
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: лог:
<--- 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=z9hG4bK557FAFFA1476067232
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>
Contact: <sip:961xxxx97@192.168.100.2>
Call-ID: 0b20-f6ac-880f-18c5@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 8352 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 - 0b20-f6ac-880f-18c5@192.168.100.2
Found peer 'SiTi_IP_M' for '961xxxx97' 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 192.168.100.2:8352
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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>
Call-ID: 0b20-f6ac-880f-18c5@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-00000aa9", "") in new stack
-- Executing [685747@from-internal:2] NoCDR("SIP/SiTi_IP_M-00000aa9", "") in new stack
-- Executing [685747@from-internal:3] Progress("SIP/SiTi_IP_M-00000aa9", "") in new stack
Audio is at 12418
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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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 703522170 703522170 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 12418 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-00000aa9", "1") in new stack
> 0x2b4b8402b7a0 -- Probation passed - setting RTP source address to 192.168.100.2:8352
-- Executing [685747@from-internal:5] Progress("SIP/SiTi_IP_M-00000aa9", "") in new stack
-- Executing [685747@from-internal:6] Playback("SIP/SiTi_IP_M-00000aa9", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/SiTi_IP_M-00000aa9> Playing 'silence/1.slin' (language 'ru')
-- <SIP/SiTi_IP_M-00000aa9> Playing 'cannot-complete-as-dialed.slin' (language 'ru')
-- <SIP/SiTi_IP_M-00000aa9> Playing 'check-number-dial-again.slin' (language 'ru')
-- Executing [685747@from-internal:7] Wait("SIP/SiTi_IP_M-00000aa9", "1") in new stack
-- Executing [685747@from-internal:8] Congestion("SIP/SiTi_IP_M-00000aa9", "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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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


<------------>
[2015-11-13 19:23:40] WARNING[10497][C-00000a32]: channel.c:4860 ast_prod: Prodding channel 'SIP/SiTi_IP_M-00000aa9' failed
== Spawn extension (from-internal, 685747, 8) exited non-zero on 'SIP/SiTi_IP_M-00000aa9'
-- Executing [h@from-internal:1] Hangup("SIP/SiTi_IP_M-00000aa9", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/SiTi_IP_M-00000aa9'
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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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 #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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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=z9hG4bK557FAFFA1476067232;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK17FE0D992145442192;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=4025250388174
To: <sip:685747@Orenburg>;tag=as48919fbb
Call-ID: 0b20-f6ac-880f-18c5@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


---
[2015-11-13 19:24:12] WARNING[4189]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 0b20-f6ac-880f-18c5@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
Really destroying SIP dialog '0b20-f6ac-880f-18c5@192.168.100.2' Method: INVITE
elastix*CLI> sip set debug off
в E1-SIP шлюзе указал чтобы при входящем на 685747 подменялся DID на 2800
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: лог:
elastix*CLI> sip set debug peer SiTi_IP_M_in
SIP Debugging Enabled for IP: 192.168.100.2

<--- SIP read from UDP:192.168.100.2:5070 --->
INVITE sip:2800@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK5C6930C9-963441520
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>
Contact: <sip:961xxxx97@192.168.100.2>
Call-ID: 49f6-820f-a904-387d@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 8354 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 - 49f6-820f-a904-387d@192.168.100.2
Found peer 'SiTi_IP_M' for '961xxxx97' 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 192.168.100.2:8354
Looking for 2800 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=z9hG4bK5C6930C9-963441520;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>
Call-ID: 49f6-820f-a904-387d@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:2800@192.168.100.45:5060>
Content-Length: 0


<------------>
-- Executing [2800@from-internal:1] Set("SIP/SiTi_IP_M-00000aac", "__RINGTIMER=15") in new stack
-- Executing [2800@from-internal:2] Macro("SIP/SiTi_IP_M-00000aac", "exten-vm,novm,2800,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/SiTi_IP_M-00000aac", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/SiTi_IP_M-00000aac", "TOUCH_MONITOR=1447425014.7628") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/SiTi_IP_M-00000aac", "AMPUSER=961xxxx97") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/SiTi_IP_M-00000aac", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/SiTi_IP_M-00000aac", "1?Set(REALCALLERIDNUM=961xxxx97)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/SiTi_IP_M-00000aac", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/SiTi_IP_M-00000aac", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/SiTi_IP_M-00000aac", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/SiTi_IP_M-00000aac", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/SiTi_IP_M-00000aac", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/SiTi_IP_M-00000aac", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/SiTi_IP_M-00000aac", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/SiTi_IP_M-00000aac", "CALLERID(number)=961xxxx97") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/SiTi_IP_M-00000aac", "CALLERID(name)=") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/SiTi_IP_M-00000aac", "CDR(cnum)=961xxxx97") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/SiTi_IP_M-00000aac", "CDR(cnam)=") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/SiTi_IP_M-00000aac", "CHANNEL(language)=ru") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/SiTi_IP_M-00000aac", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/SiTi_IP_M-00000aac", "__EXTTOCALL=2800") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/SiTi_IP_M-00000aac", "__PICKUPMARK=2800") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/SiTi_IP_M-00000aac", "RT=") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/SiTi_IP_M-00000aac", "sub-record-check,s,1(exten,2800,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/SiTi_IP_M-00000aac", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/SiTi_IP_M-00000aac", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/SiTi_IP_M-00000aac", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/SiTi_IP_M-00000aac", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/SiTi_IP_M-00000aac", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/SiTi_IP_M-00000aac", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/SiTi_IP_M-00000aac", "NOW=1447425014") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/SiTi_IP_M-00000aac", "__DAY=13") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/SiTi_IP_M-00000aac", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/SiTi_IP_M-00000aac", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/SiTi_IP_M-00000aac", "__TIMESTR=20151113-193014") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/SiTi_IP_M-00000aac", "__FROMEXTEN=961xxxx97") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/SiTi_IP_M-00000aac", "__CALLFILENAME=exten-2800-961xxxx97-20151113-193014-1447425014.7628") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/SiTi_IP_M-00000aac", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/SiTi_IP_M-00000aac", "0?callee") in new stack
-- Executing [exten@sub-record-check:2] Set("SIP/SiTi_IP_M-00000aac", "__REC_POLICY_MODE=always") in new stack
-- Executing [exten@sub-record-check:3] GotoIf("SIP/SiTi_IP_M-00000aac", "0?caller") in new stack
-- Executing [exten@sub-record-check:4] GotoIf("SIP/SiTi_IP_M-00000aac", "0?callee") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Set(CALLER_PRI=):Set(CALLER_PRI=0)") in new stack
-- Executing [exten@sub-record-check:6] ExecIf("SIP/SiTi_IP_M-00000aac", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [exten@sub-record-check:7] GotoIf("SIP/SiTi_IP_M-00000aac", "0?caller:callee") in new stack
-- Goto (sub-record-check,exten,8)
-- Executing [exten@sub-record-check:8] GosubIf("SIP/SiTi_IP_M-00000aac", "1?record,1(exten,2800,961xxxx97)") in new stack
-- Executing [record@sub-record-check:1] Set("SIP/SiTi_IP_M-00000aac", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [record@sub-record-check:2] MixMonitor("SIP/SiTi_IP_M-00000aac", "2015/11/13/exten-2800-961xxxx97-20151113-193014-1447425014.7628.wav,,") in new stack
-- Executing [record@sub-record-check:3] Set("SIP/SiTi_IP_M-00000aac", "__REC_STATUS=RECORDING") in new stack
-- Executing [record@sub-record-check:4] Set("SIP/SiTi_IP_M-00000aac", "CDR(recordingfile)=exten-2800-961xxxx97-20151113-193014-1447425014.7628.wav") in new stack
-- Executing [record@sub-record-check:5] Return("SIP/SiTi_IP_M-00000aac", "") in new stack
-- Executing [exten@sub-record-check:9] Return("SIP/SiTi_IP_M-00000aac", "") in new stack
-- Executing [s@macro-exten-vm:7] Macro("SIP/SiTi_IP_M-00000aac", "dial-one,,Ttr,2800") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000aac", "DEXTEN=2800") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000aac", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/SiTi_IP_M-00000aac", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/SiTi_IP_M-00000aac", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/SiTi_IP_M-00000aac", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/SiTi_IP_M-00000aac", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000aac", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000aac", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000aac", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/SiTi_IP_M-00000aac", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/SiTi_IP_M-00000aac", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/SiTi_IP_M-00000aac", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000aac", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000aac", "DEVICES=2800") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Set(DEVICES=5622800)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000aac", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/SiTi_IP_M-00000aac", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000aac", "THISDIAL=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/SiTi_IP_M-00000aac", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000aac", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/SiTi_IP_M-00000aac", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/SiTi_IP_M-00000aac", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000aac", "THISPART2=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Set(THISPART2=DAHDI/2800)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000aac", "NEWDIAL=SIP/2800&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/SiTi_IP_M-00000aac", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000aac", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000aac", "THISDIAL=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/SiTi_IP_M-00000aac", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/SiTi_IP_M-00000aac", "DSTRING=SIP/2800&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000aac", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000aac", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/SiTi_IP_M-00000aac", "DSTRING=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/SiTi_IP_M-00000aac", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/SiTi_IP_M-00000aac", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/SiTi_IP_M-00000aac", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/SiTi_IP_M-00000aac", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000aac", "DB(CALLTRACE/2800)=961xxxx97") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/SiTi_IP_M-00000aac", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/SiTi_IP_M-00000aac", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/SiTi_IP_M-00000aac", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/SiTi_IP_M-00000aac", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/SiTi_IP_M-00000aac", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/SiTi_IP_M-00000aac", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/SiTi_IP_M-00000aac", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/SiTi_IP_M-00000aac", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/SiTi_IP_M-00000aac", "0?usegoto,1") in new stack
== Begin MixMonitor Recording SIP/SiTi_IP_M-00000aac
-- Executing [s@macro-dial-one:38] GotoIf("SIP/SiTi_IP_M-00000aac", "1?godial") in new stack
-- Goto (macro-dial-one,s,43)
-- Executing [s@macro-dial-one:43] Dial("SIP/SiTi_IP_M-00000aac", "SIP/2800,,Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2800

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK5C6930C9-963441520;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
Call-ID: 49f6-820f-a904-387d@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:2800@192.168.100.45:5060>
Content-Length: 0


<------------>
-- SIP/2800-00000aad is ringing

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK5C6930C9-963441520;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
Call-ID: 49f6-820f-a904-387d@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:2800@192.168.100.45:5060>
Content-Length: 0


<------------>
-- SIP/2800-00000aad answered SIP/SiTi_IP_M-00000aac
Audio is at 11726
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK5C6930C9-963441520;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
Call-ID: 49f6-820f-a904-387d@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:2800@192.168.100.45:5060>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 418930168 418930168 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11726 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

<------------>
> 0x2b4b5c052780 -- Probation passed - setting RTP source address to 192.168.100.58:16480

<--- SIP read from UDP:192.168.100.2:5070 --->
ACK sip:2800@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK5C6930C9-963441520
Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bK6AAAB1FC-1431625792;received=192.168.100.2
From: <sip:961xxxx97@Orenburg>;tag=17895710258178
To: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
Contact: <sip:961xxxx97@192.168.100.2>
Call-ID: 49f6-820f-a904-387d@192.168.100.2
CSeq: 101 ACK
Expires: 180
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
> 0x2b4b8402b7a0 -- Probation passed - setting RTP source address to 192.168.100.2:8354
-- Executing [h@macro-dial-one:1] Macro("SIP/SiTi_IP_M-00000aac", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/SiTi_IP_M-00000aac", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/SiTi_IP_M-00000aac", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/SiTi_IP_M-00000aac", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/SiTi_IP_M-00000aac", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/SiTi_IP_M-00000aac", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/SiTi_IP_M-00000aac", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/SiTi_IP_M-00000aac", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/SiTi_IP_M-00000aac", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/SiTi_IP_M-00000aac", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/SiTi_IP_M-00000aac", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/SiTi_IP_M-00000aac", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/SiTi_IP_M-00000aac", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/SiTi_IP_M-00000aac>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/SiTi_IP_M-00000aac", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/SiTi_IP_M-00000aac' in macro 'hangupcall'
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/SiTi_IP_M-00000aac'
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/SiTi_IP_M-00000aac' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/SiTi_IP_M-00000aac' in macro 'exten-vm'
== Spawn extension (from-internal, 2800, 2) exited non-zero on 'SIP/SiTi_IP_M-00000aac'
Scheduling destruction of SIP dialog '49f6-820f-a904-387d@192.168.100.2' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:192.168.100.2:5070;lr> for address/port to send to
set_destination: set destination to 192.168.100.2:5070
Reliably Transmitting (no NAT) to 192.168.100.2:5070:
BYE sip:961xxxx97@192.168.100.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK2c58ab24;rport
Route: <sip:192.168.100.2:5070;lr>
Max-Forwards: 70
From: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
To: <sip:961xxxx97@Orenburg>;tag=17895710258178
Call-ID: 49f6-820f-a904-387d@192.168.100.2
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.13.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/SiTi_IP_M-00000aac

<--- SIP read from UDP:192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK2c58ab24;rport
Route: <sip:192.168.100.2:5070;lr>
Max-Forwards: 70
From: <sip:2800@elastix.56.to.fskn>;tag=as3077b4fb
To: <sip:961xxxx97@Orenburg>;tag=17895710258178
Call-ID: 49f6-820f-a904-387d@192.168.100.2
CSeq: 102 BYE
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
Server: CTBFv1.0

<------------->
--- (12 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '49f6-820f-a904-387d@192.168.100.2' Method: ACK
elastix*CLI> sip set debug off
SIP Debugging Disabled
Что вижу необычного:
1. Если прилетает городской номер, то * не находит куда его пристроить и говорит "503"
2. Появляется сообщение "183"

3. Если прилетает сразу внутренний номер, то "183" сообщения нет и отрабатывает нормально при установке любого контекса в транке (internal, pstn, trunk)

4. В логах все время "from-internal"

5. Сообщение "183" как раз-таки появилось когда щелкнул галочку "Signal RINGING"
Пробовал убирать и ставить эту галочку сызнова - эффекта ноль.

На какие мысли меня это должно навести? :)
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: Проблема Retransmission в одной подсети Elastix

Сообщение ded »

Не могу проводить техучёбу для Вас в рамках дискуссии на форуме.
Кратко: если кто-то своей рукой напишет любую глупость в extensions.conf

Код: Выделить всё

[from-internal]
include => from-internal-noxfer
include => from-internal-xfer
include => bad-number ; auto-generated
include => from-mama
include => from-Barak-Obama
Вы это примете тоже за Абсолютную истину? Строка include => from-trunk там приписана чьей то недобросовестной рукой. Поэтому такая ерунда -
Если прилетает городской номер, то * не находит куда его пристроить и говорит "503"
Потому что городской номер должен быть прописан как DID во входящей маршрутизации и там направлен на внутренний номер, или IVR, или ещё куда. И тогда он обрабатывается в контексте from-trunk. Засовывая всё в один контекст from-internal вы в одном шаге от того, чтобы создать петлю - входящий-исходящий-входящий-исходящий. Если из контекста from-internal есть исходящие наборы 6-тизначных городских типа 685747 - становится понятным почему вдруг шлюз перестаёт отвечать и бегут retransmissions. У вас всё что со шлюза прибегает - туда же и убегает сразу.
Меня просто подташнивает азбуку объяснять, Вы уж извините.

Поставить галочку, посмотреть что получится - убрать галочку, опять посмотреть. Это творческая работа укладывается в общий список - смотреть за hicom, avaya, СиТи, *, администрировать сеть, прокладывать видеонаблюдение, ремонтировать сигнализацию и заниматься бумажной работой.
Давайте я свои обязанности и круг деятельности тут опишу? Верю, что бумажной работой и сигнализацией у Вас получается заниматься лучше.

Успехов!
Вернуться к началу
RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

ded, Вы как всегда строги в своих высказываниях :) Давайте не будем мерится кругом обязанностей :) Сразу же поставил меточку что это offtop - т.е. не относящееся к конкретной теме обсуждения, просто в кругу друзей иногда можно позволить объяснить свою забывчивость или усталость, или просто о чём-то высказаться
А с бумагами может у меня ещё хуже :)

А теперь по теме, в extensions.conf в самом его начале написано о том, что этот файл генерируется автоматически и все изменения в нем затираются. Не спорю, быть может когда начинал знакомится с elastix'ом может и поставил туда данный include, но запамятовал. И, по все видимости, не все изменения в нем затираются. Спасибо что подсказали :)

Удалил данную строку, поведение при входящем городском не изменилось. Контексты в транк пробовал указывать разные и from-pstn и from-trunk и from-internal, в логах почему-то каждый раз показывает этапы [from-internal], идут гудки вместо голоса и ошибка Retransmission. Складывается устойчивое ощущение, что эти изменения не воспринимаются

Inbound DID настроен через веб-интерфейс.
Шестизначных исходящих маршрутов нет и не было.

С контекстом from-internal был лишь один раз эксперимент.

Прошло часа полтора :)
В итоге в настройках транка в секции "outgoing settings" была строка context=from-internal
По всей видимости секция outgoing setting приоритетнее секции incoming settings, так как изменив в первой строку на from-trunk, у меня заиграло :)
Но проблема с 32 секундами и Retransmission осталась. Что ещё можно проверить?
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Сип-дебаг:
elastix*CLI> sip set debug peer SiTi_IP_M_in
SIP Debugging Enabled for IP: 192.168.100.2

<--- 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=z9hG4bK795C6A81-1782142960
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>
Contact: <sip:909xxxx959@192.168.100.2>
Call-ID: 8f79-ba6b-1249-bdb0@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 8376 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 - 8f79-ba6b-1249-bdb0@192.168.100.2
Found peer 'SiTi_IP_M' for '909xxxx959' 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 192.168.100.2:8376
Looking for 685747 in from-trunk-sip-SiTi_IP_M (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=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>
Call-ID: 8f79-ba6b-1249-bdb0@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-trunk-sip-SiTi_IP_M:1] Set("SIP/SiTi_IP_M-00000012", "GROUP()=OUT_4") in new stack
-- Executing [685747@from-trunk-sip-SiTi_IP_M:2] Goto("SIP/SiTi_IP_M-00000012", "from-trunk,685747,1") in new stack
-- Goto (from-trunk,685747,1)
-- Executing [685747@from-trunk:1] Set("SIP/SiTi_IP_M-00000012", "__FROM_DID=685747") in new stack
-- Executing [685747@from-trunk:2] Gosub("SIP/SiTi_IP_M-00000012", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/SiTi_IP_M-00000012", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/SiTi_IP_M-00000012", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/SiTi_IP_M-00000012", "") in new stack
-- Executing [685747@from-trunk:3] Set("SIP/SiTi_IP_M-00000012", "__REC_POLICY_MODE=always") in new stack
-- Executing [685747@from-trunk:4] Set("SIP/SiTi_IP_M-00000012", "CDR(did)=685747") in new stack
-- Executing [685747@from-trunk:5] ExecIf("SIP/SiTi_IP_M-00000012", "1 ?Set(CALLERID(name)=909xxxx959)") in new stack
-- Executing [685747@from-trunk:6] Set("SIP/SiTi_IP_M-00000012", "CHANNEL(musicclass)=default") in new stack
-- Executing [685747@from-trunk:7] Set("SIP/SiTi_IP_M-00000012", "__MOHCLASS=default") in new stack
-- Executing [685747@from-trunk:8] Set("SIP/SiTi_IP_M-00000012", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [685747@from-trunk:9] Set("SIP/SiTi_IP_M-00000012", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [685747@from-trunk:10] Goto("SIP/SiTi_IP_M-00000012", "app-announcement-5,s,1") in new stack
-- Goto (app-announcement-5,s,1)
-- Executing [s@app-announcement-5:1] GotoIf("SIP/SiTi_IP_M-00000012", "0?begin") in new stack
-- Executing [s@app-announcement-5:2] Answer("SIP/SiTi_IP_M-00000012", "") in new stack
Audio is at 11022
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

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

<--- SIP read from UDP:192.168.100.2:5070 --->
ACK sip:685747@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK795C6A81-1782142960
Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@elastix.56.to.fskn>
Contact: <sip:909xxxx959@192.168.100.2>
Call-ID: 8f79-ba6b-1249-bdb0@192.168.100.2
CSeq: 101 ACK
Expires: 180
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
> 0x85bd7b0 -- Probation passed - setting RTP source address to 192.168.100.2:8376
-- Executing [s@app-announcement-5:3] Wait("SIP/SiTi_IP_M-00000012", "1") in new stack
Retransmitting #1 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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 [s@app-announcement-5:4] NoOp("SIP/SiTi_IP_M-00000012", "Playing announcement vh_dov") in new stack
-- Executing [s@app-announcement-5:5] Playback("SIP/SiTi_IP_M-00000012", "custom/tel_dov,noanswer") in new stack
-- <SIP/SiTi_IP_M-00000012> Playing 'custom/tel_dov.slin' (language 'ru')
Retransmitting #2 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #3 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #4 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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 [s@app-announcement-5:6] Goto("SIP/SiTi_IP_M-00000012", "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-00000012", "__RINGTIMER=15") in new stack
-- Executing [2800@from-did-direct:2] Macro("SIP/SiTi_IP_M-00000012", "exten-vm,novm,2800,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/SiTi_IP_M-00000012", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/SiTi_IP_M-00000012", "TOUCH_MONITOR=1447439332.18") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/SiTi_IP_M-00000012", "AMPUSER=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/SiTi_IP_M-00000012", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/SiTi_IP_M-00000012", "1?Set(REALCALLERIDNUM=909xxxx959)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/SiTi_IP_M-00000012", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/SiTi_IP_M-00000012", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/SiTi_IP_M-00000012", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/SiTi_IP_M-00000012", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/SiTi_IP_M-00000012", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/SiTi_IP_M-00000012", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/SiTi_IP_M-00000012", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/SiTi_IP_M-00000012", "CALLERID(number)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/SiTi_IP_M-00000012", "CALLERID(name)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/SiTi_IP_M-00000012", "CDR(cnum)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/SiTi_IP_M-00000012", "CDR(cnam)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/SiTi_IP_M-00000012", "CHANNEL(language)=ru") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/SiTi_IP_M-00000012", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/SiTi_IP_M-00000012", "__EXTTOCALL=2800") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/SiTi_IP_M-00000012", "__PICKUPMARK=2800") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/SiTi_IP_M-00000012", "RT=") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/SiTi_IP_M-00000012", "sub-record-check,s,1(exten,2800,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/SiTi_IP_M-00000012", "REC_POLICY_MODE_SAVE=always") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/SiTi_IP_M-00000012", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/SiTi_IP_M-00000012", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/SiTi_IP_M-00000012", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/SiTi_IP_M-00000012", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/SiTi_IP_M-00000012", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/SiTi_IP_M-00000012", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/SiTi_IP_M-00000012", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/SiTi_IP_M-00000012", "NOW=1447439341") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/SiTi_IP_M-00000012", "__DAY=13") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/SiTi_IP_M-00000012", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/SiTi_IP_M-00000012", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/SiTi_IP_M-00000012", "__TIMESTR=20151113-232901") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/SiTi_IP_M-00000012", "__FROMEXTEN=909xxxx959") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/SiTi_IP_M-00000012", "__CALLFILENAME=exten-2800-909xxxx959-20151113-232901-1447439332.18") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/SiTi_IP_M-00000012", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/SiTi_IP_M-00000012", "1?callee") in new stack
-- Goto (sub-record-check,exten,8)
-- Executing [exten@sub-record-check:8] GosubIf("SIP/SiTi_IP_M-00000012", "1?record,1(exten,2800,909xxxx959)") in new stack
-- Executing [record@sub-record-check:1] Set("SIP/SiTi_IP_M-00000012", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [record@sub-record-check:2] MixMonitor("SIP/SiTi_IP_M-00000012", "2015/11/13/exten-2800-909xxxx959-20151113-232901-1447439332.18.wav,,") in new stack
-- Executing [record@sub-record-check:3] Set("SIP/SiTi_IP_M-00000012", "__REC_STATUS=RECORDING") in new stack
-- Executing [record@sub-record-check:4] Set("SIP/SiTi_IP_M-00000012", "CDR(recordingfile)=exten-2800-909xxxx959-20151113-232901-1447439332.18.wav") in new stack
-- Executing [record@sub-record-check:5] Return("SIP/SiTi_IP_M-00000012", "") in new stack
-- Executing [exten@sub-record-check:9] Return("SIP/SiTi_IP_M-00000012", "") in new stack
-- Executing [s@macro-exten-vm:7] Macro("SIP/SiTi_IP_M-00000012", "dial-one,,Ttr,2800") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000012", "DEXTEN=2800") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000012", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/SiTi_IP_M-00000012", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/SiTi_IP_M-00000012", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/SiTi_IP_M-00000012", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/SiTi_IP_M-00000012", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000012", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000012", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000012", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/SiTi_IP_M-00000012", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/SiTi_IP_M-00000012", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/SiTi_IP_M-00000012", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000012", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000012", "DEVICES=2800") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/SiTi_IP_M-00000012", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/SiTi_IP_M-00000012", "0?Set(DEVICES=5622800)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000012", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/SiTi_IP_M-00000012", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000012", "THISDIAL=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/SiTi_IP_M-00000012", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/SiTi_IP_M-00000012", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000012", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/SiTi_IP_M-00000012", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/SiTi_IP_M-00000012", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000012", "THISPART2=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/SiTi_IP_M-00000012", "0?Set(THISPART2=DAHDI/2800)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000012", "NEWDIAL=SIP/2800&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/SiTi_IP_M-00000012", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000012", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000012", "THISDIAL=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/SiTi_IP_M-00000012", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/SiTi_IP_M-00000012", "DSTRING=SIP/2800&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000012", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000012", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/SiTi_IP_M-00000012", "DSTRING=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/SiTi_IP_M-00000012", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/SiTi_IP_M-00000012", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/SiTi_IP_M-00000012", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/SiTi_IP_M-00000012", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000012", "DB(CALLTRACE/2800)=909xxxx959") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/SiTi_IP_M-00000012", "") in new stack
== Begin MixMonitor Recording SIP/SiTi_IP_M-00000012
-- Executing [s@macro-dial-one:30] Set("SIP/SiTi_IP_M-00000012", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/SiTi_IP_M-00000012", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/SiTi_IP_M-00000012", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/SiTi_IP_M-00000012", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/SiTi_IP_M-00000012", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/SiTi_IP_M-00000012", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/SiTi_IP_M-00000012", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/SiTi_IP_M-00000012", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:38] GotoIf("SIP/SiTi_IP_M-00000012", "1?godial") in new stack
-- Goto (macro-dial-one,s,43)
-- Executing [s@macro-dial-one:43] Dial("SIP/SiTi_IP_M-00000012", "SIP/2800,,Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2800
-- SIP/2800-00000013 is ringing
Retransmitting #5 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #6 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #7 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #8 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #9 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
Retransmitting #10 (no NAT) to 192.168.100.2:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK795C6A81-1782142960;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bK23BE65D21004952864;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=5996799118218
To: <sip:685747@Orenburg>;tag=as46730f07
Call-ID: 8f79-ba6b-1249-bdb0@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 149331652 149331652 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 11022 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

---
[2015-11-13 23:29:24] WARNING[3128]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 8f79-ba6b-1249-bdb0@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 23:29:24] WARNING[3128]: chan_sip.c:4053 retrans_pkt: Hanging up call 8f79-ba6b-1249-bdb0@192.168.100.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/SiTi_IP_M-00000012' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/SiTi_IP_M-00000012' in macro 'exten-vm'
== Spawn extension (from-did-direct, 2800, 2) exited non-zero on 'SIP/SiTi_IP_M-00000012'
-- Executing [h@from-did-direct:1] Macro("SIP/SiTi_IP_M-00000012", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/SiTi_IP_M-00000012", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/SiTi_IP_M-00000012", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/SiTi_IP_M-00000012", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/SiTi_IP_M-00000012", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/SiTi_IP_M-00000012", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/SiTi_IP_M-00000012", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/SiTi_IP_M-00000012", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/SiTi_IP_M-00000012", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/SiTi_IP_M-00000012", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/SiTi_IP_M-00000012", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/SiTi_IP_M-00000012", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/SiTi_IP_M-00000012", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/SiTi_IP_M-00000012>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/SiTi_IP_M-00000012", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/SiTi_IP_M-00000012' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/SiTi_IP_M-00000012'
Scheduling destruction of SIP dialog '8f79-ba6b-1249-bdb0@192.168.100.2' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:192.168.100.2:5070;lr> for address/port to send to
set_destination: set destination to 192.168.100.2:5070
Reliably Transmitting (no NAT) to 192.168.100.2:5070:
BYE sip:909xxxx959@192.168.100.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK08a27d69;rport
Route: <sip:192.168.100.2:5070;lr>
Max-Forwards: 70
From: <sip:685747@Orenburg>;tag=as46730f07
To: <sip:909xxxx959@Orenburg>;tag=5996799118218
Call-ID: 8f79-ba6b-1249-bdb0@192.168.100.2
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.13.0)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/SiTi_IP_M-00000012

<--- SIP read from UDP:192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.45:5060;branch=z9hG4bK08a27d69;rport
Route: <sip:192.168.100.2:5070;lr>
Max-Forwards: 70
From: <sip:685747@Orenburg>;tag=as46730f07
To: <sip:909xxxx959@Orenburg>;tag=5996799118218
Call-ID: 8f79-ba6b-1249-bdb0@192.168.100.2
CSeq: 102 BYE
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
Server: CTBFv1.0

<------------->
--- (12 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '8f79-ba6b-1249-bdb0@192.168.100.2' Method: INVITE
elastix*CLI> sip set debug off
SIP Debugging Disabled
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Вернуться к началу
RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

Если же этот же DID навести сразу на extension 2800, то проблем нет никаких
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: SIP Debug DID to extension
elastix*CLI> sip set debug peer SiTi_IP_M_in
SIP Debugging Enabled for IP: 192.168.100.2

<--- 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=z9hG4bK58BD2641-1949146096
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@Orenburg>
Contact: <sip:909xxxx959@192.168.100.2>
Call-ID: cce0-54b2-ab4f-c6cb@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 8366 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 - cce0-54b2-ab4f-c6cb@192.168.100.2
Found peer 'SiTi_IP_M' for '909xxxx959' 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 192.168.100.2:8366
Looking for 685747 in from-trunk-sip-SiTi_IP_M (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=z9hG4bK58BD2641-1949146096;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@Orenburg>
Call-ID: cce0-54b2-ab4f-c6cb@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-trunk-sip-SiTi_IP_M:1] Set("SIP/SiTi_IP_M-00000016", "GROUP()=OUT_4") in new stack
-- Executing [685747@from-trunk-sip-SiTi_IP_M:2] Goto("SIP/SiTi_IP_M-00000016", "from-trunk,685747,1") in new stack
-- Goto (from-trunk,685747,1)
-- Executing [685747@from-trunk:1] Set("SIP/SiTi_IP_M-00000016", "__FROM_DID=685747") in new stack
-- Executing [685747@from-trunk:2] Gosub("SIP/SiTi_IP_M-00000016", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/SiTi_IP_M-00000016", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/SiTi_IP_M-00000016", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [685747@from-trunk:3] Set("SIP/SiTi_IP_M-00000016", "__REC_POLICY_MODE=always") in new stack
-- Executing [685747@from-trunk:4] Set("SIP/SiTi_IP_M-00000016", "CDR(did)=685747") in new stack
-- Executing [685747@from-trunk:5] ExecIf("SIP/SiTi_IP_M-00000016", "1 ?Set(CALLERID(name)=909xxxx959)") in new stack
-- Executing [685747@from-trunk:6] Set("SIP/SiTi_IP_M-00000016", "CHANNEL(musicclass)=default") in new stack
-- Executing [685747@from-trunk:7] Set("SIP/SiTi_IP_M-00000016", "__MOHCLASS=default") in new stack
-- Executing [685747@from-trunk:8] Set("SIP/SiTi_IP_M-00000016", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [685747@from-trunk:9] Set("SIP/SiTi_IP_M-00000016", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [685747@from-trunk:10] Goto("SIP/SiTi_IP_M-00000016", "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-00000016", "__RINGTIMER=15") in new stack
-- Executing [2800@from-did-direct:2] Macro("SIP/SiTi_IP_M-00000016", "exten-vm,novm,2800,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/SiTi_IP_M-00000016", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/SiTi_IP_M-00000016", "TOUCH_MONITOR=1447440403.22") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/SiTi_IP_M-00000016", "AMPUSER=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/SiTi_IP_M-00000016", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/SiTi_IP_M-00000016", "1?Set(REALCALLERIDNUM=909xxxx959)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/SiTi_IP_M-00000016", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/SiTi_IP_M-00000016", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/SiTi_IP_M-00000016", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/SiTi_IP_M-00000016", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/SiTi_IP_M-00000016", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/SiTi_IP_M-00000016", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/SiTi_IP_M-00000016", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/SiTi_IP_M-00000016", "CALLERID(number)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/SiTi_IP_M-00000016", "CALLERID(name)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/SiTi_IP_M-00000016", "CDR(cnum)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/SiTi_IP_M-00000016", "CDR(cnam)=909xxxx959") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/SiTi_IP_M-00000016", "CHANNEL(language)=ru") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/SiTi_IP_M-00000016", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/SiTi_IP_M-00000016", "__EXTTOCALL=2800") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/SiTi_IP_M-00000016", "__PICKUPMARK=2800") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/SiTi_IP_M-00000016", "RT=") in new stack
-- Executing [s@macro-exten-vm:6] Gosub("SIP/SiTi_IP_M-00000016", "sub-record-check,s,1(exten,2800,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/SiTi_IP_M-00000016", "REC_POLICY_MODE_SAVE=always") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/SiTi_IP_M-00000016", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/SiTi_IP_M-00000016", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/SiTi_IP_M-00000016", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/SiTi_IP_M-00000016", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/SiTi_IP_M-00000016", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/SiTi_IP_M-00000016", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/SiTi_IP_M-00000016", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/SiTi_IP_M-00000016", "NOW=1447440403") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/SiTi_IP_M-00000016", "__DAY=13") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/SiTi_IP_M-00000016", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/SiTi_IP_M-00000016", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/SiTi_IP_M-00000016", "__TIMESTR=20151113-234643") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/SiTi_IP_M-00000016", "__FROMEXTEN=909xxxx959") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/SiTi_IP_M-00000016", "__CALLFILENAME=exten-2800-909xxxx959-20151113-234643-1447440403.22") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/SiTi_IP_M-00000016", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/SiTi_IP_M-00000016", "1?callee") in new stack
-- Goto (sub-record-check,exten,8)
-- Executing [exten@sub-record-check:8] GosubIf("SIP/SiTi_IP_M-00000016", "1?record,1(exten,2800,909xxxx959)") in new stack
-- Executing [record@sub-record-check:1] Set("SIP/SiTi_IP_M-00000016", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [record@sub-record-check:2] MixMonitor("SIP/SiTi_IP_M-00000016", "2015/11/13/exten-2800-909xxxx959-20151113-234643-1447440403.22.wav,,") in new stack
-- Executing [record@sub-record-check:3] Set("SIP/SiTi_IP_M-00000016", "__REC_STATUS=RECORDING") in new stack
-- Executing [record@sub-record-check:4] Set("SIP/SiTi_IP_M-00000016", "CDR(recordingfile)=exten-2800-909xxxx959-20151113-234643-1447440403.22.wav") in new stack
-- Executing [record@sub-record-check:5] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [exten@sub-record-check:9] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [s@macro-exten-vm:7] Macro("SIP/SiTi_IP_M-00000016", "dial-one,,Ttr,2800") in new stack
== Begin MixMonitor Recording SIP/SiTi_IP_M-00000016
-- Executing [s@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000016", "DEXTEN=2800") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000016", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/SiTi_IP_M-00000016", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/SiTi_IP_M-00000016", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/SiTi_IP_M-00000016", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/SiTi_IP_M-00000016", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000016", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000016", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000016", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/SiTi_IP_M-00000016", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/SiTi_IP_M-00000016", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/SiTi_IP_M-00000016", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000016", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000016", "DEVICES=2800") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/SiTi_IP_M-00000016", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/SiTi_IP_M-00000016", "0?Set(DEVICES=5622800)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000016", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/SiTi_IP_M-00000016", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000016", "THISDIAL=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/SiTi_IP_M-00000016", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/SiTi_IP_M-00000016", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/SiTi_IP_M-00000016", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/SiTi_IP_M-00000016", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/SiTi_IP_M-00000016", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/SiTi_IP_M-00000016", "THISPART2=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/SiTi_IP_M-00000016", "0?Set(THISPART2=DAHDI/2800)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/SiTi_IP_M-00000016", "NEWDIAL=SIP/2800&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/SiTi_IP_M-00000016", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/SiTi_IP_M-00000016", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000016", "THISDIAL=SIP/2800") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/SiTi_IP_M-00000016", "DSTRING=SIP/2800&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/SiTi_IP_M-00000016", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/SiTi_IP_M-00000016", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/SiTi_IP_M-00000016", "DSTRING=SIP/2800") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/SiTi_IP_M-00000016", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/SiTi_IP_M-00000016", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/SiTi_IP_M-00000016", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/SiTi_IP_M-00000016", "DB(CALLTRACE/2800)=909xxxx959") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/SiTi_IP_M-00000016", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/SiTi_IP_M-00000016", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/SiTi_IP_M-00000016", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/SiTi_IP_M-00000016", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/SiTi_IP_M-00000016", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/SiTi_IP_M-00000016", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/SiTi_IP_M-00000016", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/SiTi_IP_M-00000016", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/SiTi_IP_M-00000016", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:38] GotoIf("SIP/SiTi_IP_M-00000016", "1?godial") in new stack
-- Goto (macro-dial-one,s,43)
-- Executing [s@macro-dial-one:43] Dial("SIP/SiTi_IP_M-00000016", "SIP/2800,,Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2800

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK58BD2641-1949146096;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@Orenburg>;tag=as06fa5eb3
Call-ID: cce0-54b2-ab4f-c6cb@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


<------------>
-- SIP/2800-00000017 is ringing

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK58BD2641-1949146096;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@Orenburg>;tag=as06fa5eb3
Call-ID: cce0-54b2-ab4f-c6cb@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


<------------>
-- SIP/2800-00000017 answered SIP/SiTi_IP_M-00000016
Audio is at 15166
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK58BD2641-1949146096;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;rport=5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@Orenburg>;tag=as06fa5eb3
Call-ID: cce0-54b2-ab4f-c6cb@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 370863368 370863368 IN IP4 192.168.100.45
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.100.45
t=0 0
m=audio 15166 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

<------------>
> 0x2b096c874220 -- Probation passed - setting RTP source address to 192.168.100.58:16492

<--- SIP read from UDP:192.168.100.2:5070 --->
ACK sip:685747@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK58BD2641-1949146096
Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@elastix.56.to.fskn>;tag=as06fa5eb3
Contact: <sip:909xxxx959@192.168.100.2>
Call-ID: cce0-54b2-ab4f-c6cb@192.168.100.2
CSeq: 101 ACK
Expires: 180
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
> 0x2b09980dbff0 -- Probation passed - setting RTP source address to 192.168.100.2:8366

<--- SIP read from UDP:192.168.100.2:5070 --->
BYE sip:685747@elastix.56.to.fskn SIP/2.0
Via: SIP/2.0/UDP 192.168.100.2:5070;rport;branch=z9hG4bK58BD2641-1949146096
Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
User-Agent: CTBFv1.0
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@elastix.56.to.fskn>;tag=as06fa5eb3
Call-ID: cce0-54b2-ab4f-c6cb@192.168.100.2
CSeq: 102 BYE
Content-Length: 0
Record-Route: <sip:192.168.100.2:5070;lr>

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.100.2:5070 (no NAT)
Scheduling destruction of SIP dialog 'cce0-54b2-ab4f-c6cb@192.168.100.2' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.100.2:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.2:5070;branch=z9hG4bK58BD2641-1949146096;received=192.168.100.2;rport=5070
Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bKB7523E7-1219346832;received=192.168.100.2
Record-Route: <sip:192.168.100.2:5070;lr>
From: <sip:909xxxx959@Orenburg>;tag=1922257248220
To: <sip:685747@elastix.56.to.fskn>;tag=as06fa5eb3
Call-ID: cce0-54b2-ab4f-c6cb@192.168.100.2
CSeq: 102 BYE
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


<------------>
-- Executing [h@macro-dial-one:1] Macro("SIP/SiTi_IP_M-00000016", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/SiTi_IP_M-00000016", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/SiTi_IP_M-00000016", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/SiTi_IP_M-00000016", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/SiTi_IP_M-00000016", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/SiTi_IP_M-00000016", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/SiTi_IP_M-00000016", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/SiTi_IP_M-00000016", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/SiTi_IP_M-00000016", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/SiTi_IP_M-00000016", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/SiTi_IP_M-00000016", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/SiTi_IP_M-00000016", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/SiTi_IP_M-00000016", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/SiTi_IP_M-00000016>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/SiTi_IP_M-00000016", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/SiTi_IP_M-00000016' in macro 'hangupcall'
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/SiTi_IP_M-00000016'
== Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/SiTi_IP_M-00000016' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/SiTi_IP_M-00000016' in macro 'exten-vm'
== Spawn extension (from-did-direct, 2800, 2) exited non-zero on 'SIP/SiTi_IP_M-00000016'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/SiTi_IP_M-00000016
elastix*CLI> sip set debug off
SIP Debugging Disabled
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RUMarat
Сообщения: 18
Зарегистрирован: 06 май 2015, 11:43
Откуда: Oren

Re: Проблема Retransmission в одной подсети Elastix

Сообщение RUMarat »

Всем огромнейшее спасибо за помощь!

Вопрос закрыт галочкой "Signal RINGING" (в русскоязычном интерфейсе "Сигнализация вызова (КПВ)")

ded, спасибо что направил в нужную сторону и дал разобраться самому (так лучше запоминается :))

Всем хороших выходных!
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