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Пропадает связь через 30 сек

Проблемы и их решения Asterisk как такового

Модераторы: april22, Zavr2008

Пропадает связь через 30 сек

Сообщение freeman33 » 26 май 2021, 13:22

Приветствую!

Есть транк по VPN FreeSwitch - Asterisk

Пока есть связь тоьлко в одну сторону FreeSwitch - Asterisk, но через 30 секю соединение рвется. В интернетах пишут что скорее всего Aster отвечает не туда куда надо, я не могу понять это по логам (VoIPт -телефонией никогда не занимался). Есть лог звонка со стороны Astera, видно что он постоянно пытается отправить ответ.

Подскажите что и где надо настроить чтоб связь не прерывалась? Вторая задача настроить связь в сторону Asterisk-FreeSwitch, на четырехзначные номера.


[Показать] Спойлер:
-- PJSIP/180-00000055 is ringing
-- PJSIP/180-00000055 is ringing
<--- Transmitting SIP response (513 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Content-Length: 0


> 0x7f9230140530 -- Strict RTP learning after remote address set to: 192.168.103.198:16470
-- PJSIP/180-00000055 answered PJSIP/ekttrank-00000054
> 0x7f9230040140 -- Strict RTP learning after remote address set to: 172.18.253.1:19124
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Channel PJSIP/180-00000055 joined 'simple_bridge' basic-bridge <84285474-9134-49e3-a901-2607e279f1d5>
-- Channel PJSIP/ekttrank-00000054 joined 'simple_bridge' basic-bridge <84285474-9134-49e3-a901-2607e279f1d5>
> 0x7f9230140530 -- Strict RTP switching to RTP target address 192.168.103.198:16470 as source
> 0x7f9230040140 -- Strict RTP switching to RTP target address 172.18.253.1:19124 as source
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (600 bytes) to UDP:172.18.253.1:5060 --->
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPj90f78fa3-b494-4f5e-a4cc-41e5702088d8
From: <sip:ekttrank@192.168.103.3>;tag=aee25312-d706-46d7-8cf4-812688616350
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: f15e8ee8-1a7e-4a67-8fa7-8dd1fef66fd9
CSeq: 63981 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0


<--- Received SIP response (920 bytes) from UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPj90f78fa3-b494-4f5e-a4cc-41e5702088d8;received=192.168.103.3
From: <sip:ekttrank@192.168.103.3>;tag=aee25312-d706-46d7-8cf4-812688616350
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;tag=tg7vpF7KBK3vK
Call-ID: f15e8ee8-1a7e-4a67-8fa7-8dd1fef66fd9
CSeq: 63981 OPTIONS
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0


<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

> 0x7f9230040140 -- Strict RTP learning complete - Locking on source address 172.18.253.1:19124
> 0x7f9230140530 -- Strict RTP learning complete - Locking on source address 192.168.103.198:16470
<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (900 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bK0a54pc04pX9XH
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
CSeq: 36455794 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xx.xxx.xxx.xx:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310

v=0
o=- 1622018242 1622018245 IN IP4 xx.xxx.xxx.xx
s=Asterisk
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 12832 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (495 bytes) to UDP:172.18.253.1:5060 --->
BYE sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPj5c4429fa-447d-42a0-b3a4-749e6c8ef606
From: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
To: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
CSeq: 10510 BYE
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0


<--- Received SIP response (521 bytes) from UDP:172.18.253.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPj5c4429fa-447d-42a0-b3a4-749e6c8ef606;received=192.168.103.3
From: <sip:180@192.168.103.3>;tag=43402be3-bc41-4dfd-9cab-8bee62b5c4a2
To: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
CSeq: 10510 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0


-- Channel PJSIP/ekttrank-00000054 left 'simple_bridge' basic-bridge <84285474-9134-49e3-a901-2607e279f1d5>
== Spawn extension (macro-dial-one, s, 56) exited non-zero on 'PJSIP/ekttrank-00000054' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 26) exited non-zero on 'PJSIP/ekttrank-00000054' in macro 'exten-vm'
== Spawn extension (ext-local, 180, 3) exited non-zero on 'PJSIP/ekttrank-00000054'
-- Executing [h@ext-local:1] Macro("PJSIP/ekttrank-00000054", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/ekttrank-00000054", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/ekttrank-00000054", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("PJSIP/ekttrank-00000054", "PJSIP/180-00000055 montior file= ") in new stack
-- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/ekttrank-00000054", "1?skipagi") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("PJSIP/ekttrank-00000054", "") in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/ekttrank-00000054' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/ekttrank-00000054'
-- Channel PJSIP/180-00000055 left 'simple_bridge' basic-bridge <84285474-9134-49e3-a901-2607e279f1d5>
<--- Transmitting SIP request (597 bytes) to UDP:172.18.253.1:5060 --->
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPjb61283da-4f56-4dc7-bcf0-aa88d5b3d6f4
From: <sip:ekttrank@192.168.103.3>;tag=4ad8eb06-b37a-4e9f-9918-d2d13a618026
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: 4340fc7e-7b58-4215-81c7-f77fe134eec8
CSeq: 65 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0


PS не вываливать портянки - есть соответствующие теги ....
freeman33
 
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Re: Пропадает связь через 30 сек

Сообщение freeman33 » 26 май 2021, 14:39

я так понимаю, что в поле From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar содержится не то что надо.
Что должно быть в нем, чтобы ответ вернулся на FreeSwitch?
freeman33
 
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Re: Пропадает связь через 30 сек

Сообщение freeman33 » 26 май 2021, 15:31

+

еще накопал в логе, как я понимаю не проходит авторизация, как ее настроить?

<--- Transmitting SIP response (502 bytes) to UDP:172.18.253.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKZ1BcNHF1SmKBp
Call-ID: 9806eab3-38a2-123a-ad8f-000c29c1da69
From: "2788280" <sip:ekttrank@uraltep>;tag=S7D4mmpgeaDar
To: <sip:180@192.168.103.3>;tag=z9hG4bKZ1BcNHF1SmKBp
CSeq: 36455793 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1622019171/9c37ae349c5585b0e7918ecb1f0f8af3",opaque="4b44c4180c3043d0",algorithm=md5,qop="auth"
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
freeman33
 
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Re: Пропадает связь через 30 сек

Сообщение ded » 26 май 2021, 16:45

freeman33 писал(а):VoIPт -телефонией никогда не занимался
А сетями занимались?
mtr ip_addr_freeswitch покажет что туда маршрут по туннелю есть, и он не НАТится.
tcpdump на Астериске и на Freeswitch покажет что трафик какой-то вообще бежит между ними, или не бежит.
ded
 
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Re: Пропадает связь через 30 сек

Сообщение tma » 26 май 2021, 18:18

По трейсу похоже пакеты в одну сторону попросту не доходят.
SkyTel OU - облачная АТС, DID, SIP-транк с посекундной тарификаицей, мобильная связь
http://skytel24.com | Эстония: +372.333.55.10 | Россия: +7(495)4019900
tma
 
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Re: Пропадает связь через 30 сек

Сообщение freeman33 » 27 май 2021, 06:57

ded писал(а):
freeman33 писал(а):VoIPт -телефонией никогда не занимался
А сетями занимались?
mtr ip_addr_freeswitch покажет что туда маршрут по туннелю есть, и он не НАТится.
tcpdump на Астериске и на Freeswitch покажет что трафик какой-то вообще бежит между ними, или не бежит.


занимался. Ping между ними ходит + разговор длиться 30 секунд, как тогда получается трафик идет?
freeman33
 
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Re: Пропадает связь через 30 сек

Сообщение ded » 27 май 2021, 09:04

Ну если вы считаете что пинга между ними должно хватать - вам в раздел Бизнес, платный суппорт.
ded
 
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Re: Пропадает связь через 30 сек

Сообщение freeman33 » 27 май 2021, 10:49

я понять хочу, получается разговор между двумя абонентами в течение 30 секунд это не показатель того что трафик ходит правильно? Проверю снифером что приходит что нет.
freeman33
 
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Re: Пропадает связь через 30 сек

Сообщение Zavr2008 » 27 май 2021, 11:32

разговор между двумя абонентами в течение 30 секунд это не показатель

Логи консоли астера видите? там же ясно наверняка пишет про Critical packet retransmission..
30 секунд - стандартный таймер в SIP.
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1,Подключение к ИС "Антифрод" E1 PRI/SS#7 УВР Телестор, Грифин и др..
Аватар пользователя
Zavr2008
 
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Re: Пропадает связь через 30 сек

Сообщение freeman33 » 27 май 2021, 12:44

ded писал(а):
freeman33 писал(а):VoIPт -телефонией никогда не занимался
А сетями занимались?
mtr ip_addr_freeswitch покажет что туда маршрут по туннелю есть, и он не НАТится.
tcpdump на Астериске и на Freeswitch покажет что трафик какой-то вообще бежит между ними, или не бежит.


не натится, это плохо что ли, в чем проблема что он не натится?

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

Какой-то бежит трафик
tcpdump c asteriska
[Показать] Спойлер:
[root@opnPBX2 aleksroot]# tcpdump -nvvvi eth0 src host 172.18.253.1
tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 262144 bytes
15:23:58.850550 IP (tos 0x0, ttl 62, id 11541, offset 0, flags [none], proto UDP (17), length 948)
172.18.253.1.sip > 192.168.103.3.sip: [udp sum ok] SIP, length: 920
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPj9200cc53-382f-44e0-8057-629f5ba09a86;received=192.168.103.3
From: <sip:ekttrank@192.168.103.3>;tag=94f8e092-ef53-416c-81e8-e0674e29805d
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;tag=Zeg0jt5FQaScp
Call-ID: 9d400c28-efee-44aa-a844-ca811cde3e70
CSeq: 55382 OPTIONS
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

15:24:58.850520 IP (tos 0x0, ttl 62, id 21681, offset 0, flags [none], proto UDP (17), length 948)
172.18.253.1.sip > 192.168.103.3.sip: [udp sum ok] SIP, length: 920
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPj8e06e4d0-6809-499c-a370-03745a36421e;received=192.168.103.3
From: <sip:ekttrank@192.168.103.3>;tag=ade9c2b8-0e00-4dd0-b10d-033d3c03940b
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;tag=y8HKU45y4rXvj
Call-ID: 832a602a-c99a-440c-b5b5-419c68f655c1
CSeq: 48271 OPTIONS
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

15:25:58.850480 IP (tos 0x0, ttl 62, id 35673, offset 0, flags [none], proto UDP (17), length 948)
172.18.253.1.sip > 192.168.103.3.sip: [udp sum ok] SIP, length: 920
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport=5060;branch=z9hG4bKPje6ffc377-5eb7-40b4-9853-9c9ad57ec5d1;received=192.168.103.3
From: <sip:ekttrank@192.168.103.3>;tag=47402fd2-4152-495b-93ea-4486a093c222
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;tag=7SNjBrXya954a
Call-ID: e8ff0753-c096-4084-befc-bb36fe521bcf
CSeq: 64801 OPTIONS
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

15:26:29.792001 IP (tos 0x0, ttl 62, id 43081, offset 0, flags [none], proto UDP (17), length 936)
172.18.253.1.sip > 192.168.103.3.sip: [udp sum ok] SIP, length: 908
REGISTER sip:192.168.103.3;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.18.253.1;rport;branch=z9hG4bKmQ1DaccetD03r
Max-Forwards: 70
From: <sip:ekttrank@uraltep>;tag=j5ZQ26jDUvrNS
To: <sip:ekttrank@uraltep>
Call-ID: 22ad5719-5370-4abd-bfc1-8b41d3e622ec
CSeq: 36374096 REGISTER
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Expires: 800
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Authorization: Digest username="ekttrank", realm="asterisk", nonce="1622103203/8c780307f24f86904400b301a75263e6", cnonce="Pu5/EDlmEjqPrQAMKcHaaQ", opaque="2ad23d6e7754fd11", algorithm=MD5, uri="sip:192.168.103.3;transport=udp", response="63025943146f8a4709182725470654ea", qop=auth, nc=00000002
Content-Length: 0

15:26:29.817484 IP (tos 0x0, ttl 62, id 43083, offset 0, flags [none], proto UDP (17), length 936)
172.18.253.1.sip > 192.168.103.3.sip: [udp sum ok] SIP, length: 908
REGISTER sip:192.168.103.3;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.18.253.1;rport;branch=z9hG4bKN0t6B7vHQpppm
Max-Forwards: 70
From: <sip:ekttrank@uraltep>;tag=j5ZQ26jDUvrNS
To: <sip:ekttrank@uraltep>
Call-ID: 22ad5719-5370-4abd-bfc1-8b41d3e622ec
CSeq: 36374097 REGISTER
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Expires: 800
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Authorization: Digest username="ekttrank", realm="asterisk", nonce="1622103989/7f3b17e5fdfc7632d8ab92375bc7c4d1", cnonce="E+JW4DloEjqPrQAMKcHaaQ", opaque="124161a16d044e47", algorithm=MD5, uri="sip:192.168.103.3;transport=udp", response="60127f6ea665a758932b4ba35cfd4bc9", qop=auth, nc=00000001
Content-Length: 0

^C
5 packets captured
5 packets received by filter
0 packets dropped by kernel



tcpdump c freeswitch'a

[Показать] Спойлер:
root@pbx:/home/zaudaiv# tcpdump -nvvvi eth0 src host 192.168.103.3
tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 262144 bytes
13:23:58.838812 IP (tos 0x60, ttl 62, id 60392, offset 0, flags [DF], proto UDP (17), length 628)
192.168.103.3.5060 > 172.18.253.1.5060: [udp sum ok] SIP, length: 600
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPj9200cc53-382f-44e0-8057-629f5ba09a86
From: <sip:ekttrank@192.168.103.3>;tag=94f8e092-ef53-416c-81e8-e0674e29805d
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: 9d400c28-efee-44aa-a844-ca811cde3e70
CSeq: 55382 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

13:24:58.838720 IP (tos 0x60, ttl 62, id 14883, offset 0, flags [DF], proto UDP (17), length 628)
192.168.103.3.5060 > 172.18.253.1.5060: [udp sum ok] SIP, length: 600
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPj8e06e4d0-6809-499c-a370-03745a36421e
From: <sip:ekttrank@192.168.103.3>;tag=ade9c2b8-0e00-4dd0-b10d-033d3c03940b
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: 832a602a-c99a-440c-b5b5-419c68f655c1
CSeq: 48271 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

13:25:58.838597 IP (tos 0x60, ttl 62, id 51229, offset 0, flags [DF], proto UDP (17), length 628)
192.168.103.3.5060 > 172.18.253.1.5060: [udp sum ok] SIP, length: 600
OPTIONS sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226 SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xx:5060;rport;branch=z9hG4bKPje6ffc377-5eb7-40b4-9853-9c9ad57ec5d1
From: <sip:ekttrank@192.168.103.3>;tag=47402fd2-4152-495b-93ea-4486a093c222
To: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@172.18.253.1;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>
Contact: <sip:ekttrank@xx.xxx.xxx.xx:5060>
Call-ID: e8ff0753-c096-4084-befc-bb36fe521bcf
CSeq: 64801 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

13:26:29.805582 IP (tos 0x60, ttl 62, id 12506, offset 0, flags [DF], proto UDP (17), length 532)
192.168.103.3.5060 > 172.18.253.1.5060: [udp sum ok] SIP, length: 504
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKmQ1DaccetD03r
Call-ID: 22ad5719-5370-4abd-bfc1-8b41d3e622ec
From: <sip:ekttrank@uraltep>;tag=j5ZQ26jDUvrNS
To: <sip:ekttrank@uraltep>;tag=z9hG4bKmQ1DaccetD03r
CSeq: 36374096 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1622103989/7f3b17e5fdfc7632d8ab92375bc7c4d1",opaque="124161a16d044e47",stale=true,algorithm=md5,qop="auth"
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

13:26:29.831709 IP (tos 0x60, ttl 62, id 12508, offset 0, flags [DF], proto UDP (17), length 557)
192.168.103.3.5060 > 172.18.253.1.5060: [udp sum ok] SIP, length: 529
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.253.1;rport=5060;received=172.18.253.1;branch=z9hG4bKN0t6B7vHQpppm
Call-ID: 22ad5719-5370-4abd-bfc1-8b41d3e622ec
From: <sip:ekttrank@uraltep>;tag=j5ZQ26jDUvrNS
To: <sip:ekttrank@uraltep>;tag=z9hG4bKN0t6B7vHQpppm
CSeq: 36374097 REGISTER
Date: Thu, 27 May 2021 08:26:29 GMT
Contact: <sip:gw+d97ef6e6-33d7-4e5f-bcd6-e8487ee40226@xx.xxx.xxx.xx:5060;transport=udp;gw=d97ef6e6-33d7-4e5f-bcd6-e8487ee40226>;expires=799
Expires: 800
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

^C
5 packets captured
5 packets received by filter
0 packets dropped by kernel
freeman33
 
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