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PJSIP транк, BYE без кода причины

https://forum.asterisk.ru/viewtopic.php?t=8728

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PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 15:26
valmon
Приветствую коллеги.

Не могу понять, почему не приходит код отбоя, звонок попадает в транк, но в трубке девушка говорит что сервис недоступен, или чтото там, хотя по сигналке обмен идет нормально

Код: Выделить всё

INVITE sip:7047@10.11.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.2:5060;rport;branch=z9hG4bKPj9RuZSt4wNYrhlq1oBWIFrErCwzsOCL42
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>
Contact: <sip:asterisk@10.0.15.2:5060>
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10289 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-13.0.190.11(13.11.2)
Content-Type: application/sdp
Content-Length:   309

v=0
o=- 2071759352 2071759352 IN IP4 10.0.15.2
s=Asterisk
c=IN IP4 10.0.15.2
t=0 0
m=audio 17070 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2017-02-10 15:17:50] VERBOSE[1891] res_pjsip_logger.c: <--- Received SIP response (538 bytes) from UDP:10.11.0.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj9RuZSt4wNYrhlq1oBWIFrErCwzsOCL42;received=10.11.22.2;rport=5060
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10289 INVITE
Server: FPBX-2.10.1(1.8.10.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7047@10.11.0.4:5060>
Content-Length: 0


[2017-02-10 15:17:50] VERBOSE[1891] res_pjsip_logger.c: <--- Received SIP response (887 bytes) from UDP:10.11.0.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj9RuZSt4wNYrhlq1oBWIFrErCwzsOCL42;received=10.11.22.2;rport=5060
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>;tag=as4cd223db
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10289 INVITE
Server: FPBX-2.10.1(1.8.10.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7047@10.11.0.4:5060>
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 2084800297 2084800297 IN IP4 10.11.0.4
s=Asterisk PBX 1.8.10.1
c=IN IP4 10.11.0.4
t=0 0
m=audio 14376 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[2017-02-10 15:17:50] VERBOSE[29543] res_pjsip_logger.c: <--- Transmitting SIP request (385 bytes) to UDP:10.11.0.4:5060 --->
ACK sip:7047@10.11.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.2:5060;rport;branch=z9hG4bKPjvHKIDJVer0cxNbwVkRitbtMv9Gr5rsJx
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>;tag=as4cd223db
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10289 ACK
Max-Forwards: 70
User-Agent: FPBX-13.0.190.11(13.11.2)
Content-Length:  0


[2017-02-10 15:17:50] VERBOSE[20076][C-00000116] app_dial.c: PJSIP/kf-pjsip-0000003c answered SIP/9996-00000202
[2017-02-10 15:17:50] VERBOSE[20079][C-00000116] bridge_channel.c: Channel PJSIP/kf-pjsip-0000003c joined 'simple_bridge' basic-bridge <fef5ef6e-5e30-4dc4-84f2-1f16294e9102>
[2017-02-10 15:17:50] VERBOSE[20076][C-00000116] bridge_channel.c: Channel SIP/9996-00000202 joined 'simple_bridge' basic-bridge <fef5ef6e-5e30-4dc4-84f2-1f16294e9102>
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] bridge_channel.c: Channel SIP/9996-00000202 left 'simple_bridge' basic-bridge <fef5ef6e-5e30-4dc4-84f2-1f16294e9102>
[2017-02-10 15:17:53] VERBOSE[20079][C-00000116] bridge_channel.c: Channel PJSIP/kf-pjsip-0000003c left 'simple_bridge' basic-bridge <fef5ef6e-5e30-4dc4-84f2-1f16294e9102>
[2017-02-10 15:17:53] VERBOSE[8770] res_pjsip_logger.c: <--- Transmitting SIP request (409 bytes) to UDP:10.11.0.4:5060 --->
BYE sip:7047@10.11.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.2:5060;rport;branch=z9hG4bKPj2JDBfy7K0YFhN7bc3Vo-dNauHzwW6QfS
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>;tag=as4cd223db
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10290 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-13.0.190.11(13.11.2)
Content-Length:  0


[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/9996-00000202' in macro 'dialout-trunk'
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Spawn extension (from-internal, 7047, 6) exited non-zero on 'SIP/9996-00000202'
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Executing [h@from-internal:1] Macro("SIP/9996-00000202", "hangupcall") in new stack
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/9996-00000202", "1?theend") in new stack
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/9996-00000202", "0?Set(CDR(recordingfile)=)") in new stack
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/9996-00000202", "") in new stack
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/9996-00000202' in macro 'hangupcall'
[2017-02-10 15:17:53] VERBOSE[20076][C-00000116] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/9996-00000202'
[2017-02-10 15:17:53] VERBOSE[20077][C-00000116] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2017-02-10 15:17:53] VERBOSE[20077][C-00000116] app_mixmonitor.c: End MixMonitor Recording SIP/9996-00000202
[2017-02-10 15:17:53] VERBOSE[1891] res_pjsip_logger.c: <--- Received SIP response (473 bytes) from UDP:10.11.0.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj2JDBfy7K0YFhN7bc3Vo-dNauHzwW6QfS;received=10.11.22.2;rport=5060
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=MgTugmL62HG8gYU.sD5NesyG8927xaqq
To: <sip:7047@10.11.0.4>;tag=as4cd223db
Call-ID: 1F6tBfyPqBMjOnsp8nO-KSJ-stwhYnNQ
CSeq: 10290 BYE
Server: FPBX-2.10.1(1.8.10.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 15:27
valmon
Смущает в ответе с той стороны ip адрес нашего openvpn сервера, для этого транка
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj2JDBfy7K0YFhN7bc3Vo-dNauHzwW6QfS;received=10.11.22.2;rport=5060

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 15:31
valmon
Забыл добавить, с той стороны все пролетает отлично.

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 15:56
valmon
И еще

Код: Выделить всё

[2017-02-10 15:52:00] WARNING[8770] res_pjsip_outbound_registration.c: Fatal response '403' received from 'sip:10.11.0.4:5060' on registration attempt to 'sip:jetlogin@10.11.0.4:5060', stopping outbound registration

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 16:03
ded
Может быть при некорректно работающем VPN, при котором трассировка туннелируется в одну сторону, а в противоположную - бежит поверх туннеля (и/или НАТ)..
На уровне SIP разрыв понятен, происходит BYE из-за того, что SIP/9996 покидает бридж
Channel SIP/9996-00000202 left 'simple_bridge' basic-bridge

А он может это делать по причине отсутствия RTP в одну сторону. То есть проблема неправильной маршрутизации порождает плановый разрыв соединения. Это можно задампить или tcdump или
rtp set debug on.
А в чём смысл использования именно PJSIP?

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 16:48
valmon
Да судя по всему трафик бегает

Код: Выделить всё

16:46:53.760438 IP 10.0.15.2.5060 > 10.11.0.4.5060: SIP, length: 953
16:46:53.784544 IP 10.11.0.4.5060 > 10.0.15.2.5060: SIP, length: 538
16:46:53.784881 IP 10.11.0.4.5060 > 10.0.15.2.5060: SIP, length: 887
16:46:53.785159 IP 10.0.15.2.5060 > 10.11.0.4.5060: SIP, length: 385
16:46:53.975486 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:53.994696 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:54.013805 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:54.034237 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:54.053278 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:54.072434 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:46:54.096603 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172

Код: Выделить всё

16:47:00.318728 IP 10.11.0.4.14518 > 10.0.15.2.17944: UDP, length 172
16:47:00.329360 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:47:00.339457 IP 10.11.0.4.14518 > 10.0.15.2.17944: UDP, length 172
16:47:00.353244 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:47:00.358559 IP 10.11.0.4.14518 > 10.0.15.2.17944: UDP, length 172
16:47:00.371727 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
16:47:00.378950 IP 10.11.0.4.14518 > 10.0.15.2.17944: UDP, length 172
16:47:00.390433 IP 10.0.15.2.17944 > 10.11.0.4.14518: UDP, length 172
^C1232 packets captured
1234 packets received by filter

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 17:15
ded
ded писал(а):А в чём смысл использования именно PJSIP?

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 17:35
valmon
Это не мой конфиг, вот и приходится ковырять существующий.

А в чем принципиальная разница?

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 17:41
ded
Ну, у Вас два Астериска, Вы же видите синтаксис chan_sip и chan_pjsip ??
PJSIP сложней, меньше отлажен, больше случаев непонятного поведения (как у Вас сейчас). Если в производство - то лучше chan_sip.
Если интерес поэкспериментировать - то chan_pjsip. Но тогда общаться и вопросы задавать - к разработчикам.

Re: PJSIP транк, BYE без кода причины

Добавлено: 10 фев 2017, 17:47
valmon
Тоже верно.

Все мне покоя не дает ip openvpn сервера, откуда он взялся.

<--- SIP read from UDP:10.11.22.2:5060 ---> пакета должны прилетать с 10.0.15.2, сеть маршрутизируемая, нат там нет

Это лог с той стороны

Код: Выделить всё

<--- SIP read from UDP:10.11.22.2:5060 --->
INVITE sip:7069@10.11.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.2:5060;rport;branch=z9hG4bKPj3bY0XZqdNacOR6F7SYkTY07IAETIwaXI
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=7z6l2V9qb4PqdEq0QW70miTF9BMxiVUo
To: <sip:7069@10.11.0.4>
Contact: <sip:asterisk@10.0.15.2:5060>
Call-ID: uWqt-YVN9zAF5x7eBB4Ad6TKkITUMzau
CSeq: 23084 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-13.0.190.11(13.11.2)
Content-Type: application/sdp
Content-Length: 307

v=0
o=- 133325768 133325768 IN IP4 10.0.15.2
s=Asterisk
c=IN IP4 10.0.15.2
t=0 0
m=audio 15268 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: --- (15 headers 15 lines) ---
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Sending to 10.11.22.2:5060 (NAT)
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Using INVITE request as basis request - uWqt-YVN9zAF5x7eBB4Ad6TKkITUMzau
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: No matching peer for '9996' from '10.11.22.2:5060'
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found RTP audio format 0
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found RTP audio format 8
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found RTP audio format 3
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found RTP audio format 111
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found RTP audio format 101
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found audio description format PCMU for ID 0
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found audio description format PCMA for ID 8
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found audio description format GSM for ID 3
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found audio description format G726-32 for ID 111
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Found audio description format telephone-event for ID 101
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Capabilities: us - 0xf (g723|gsm|ulaw|alaw), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Peer audio RTP is at port 10.0.15.2:15268
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: Looking for 7069 in from-sip-external (domain 10.11.0.4)
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: list_route: hop: <sip:asterisk@10.0.15.2:5060>
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: 
<--- Transmitting (NAT) to 10.11.22.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj3bY0XZqdNacOR6F7SYkTY07IAETIwaXI;received=10.11.22.2;rport=5060
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=7z6l2V9qb4PqdEq0QW70miTF9BMxiVUo
To: <sip:7069@10.11.0.4>
Call-ID: uWqt-YVN9zAF5x7eBB4Ad6TKkITUMzau
CSeq: 23084 INVITE
Server: FPBX-2.10.1(1.8.10.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7069@10.11.0.4:5060>
Content-Length: 0


<------------>
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: Audio is at 17660
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2017-02-10 11:42:31] VERBOSE[20672] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 10.11.22.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.15.2:5060;branch=z9hG4bKPj3bY0XZqdNacOR6F7SYkTY07IAETIwaXI;received=10.11.22.2;rport=5060
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=7z6l2V9qb4PqdEq0QW70miTF9BMxiVUo
To: <sip:7069@10.11.0.4>;tag=as165ba21c
Call-ID: uWqt-YVN9zAF5x7eBB4Ad6TKkITUMzau
CSeq: 23084 INVITE
Server: FPBX-2.10.1(1.8.10.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7069@10.11.0.4:5060>
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 614674805 614674805 IN IP4 10.11.0.4
s=Asterisk PBX 1.8.10.1
c=IN IP4 10.11.0.4
t=0 0
m=audio 17660 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[2017-02-10 11:42:31] VERBOSE[27835] chan_sip.c: 
<--- SIP read from UDP:10.11.22.2:5060 --->
ACK sip:7069@10.11.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.15.2:5060;rport;branch=z9hG4bKPjhNqTmucqzp2Dp2zBiyoZhVjmmTpxOPSA
From: "Dmitrij Ivanov" <sip:9996@10.0.15.2>;tag=7z6l2V9qb4PqdEq0QW70miTF9BMxiVUo
To: <sip:7069@10.11.0.4>;tag=as165ba21c
Call-ID: uWqt-YVN9zAF5x7eBB4Ad6TKkITUMzau
CSeq: 23084 ACK
Max-Forwards: 70
User-Agent: FPBX-13.0.190.11(13.11.2)
Content-Length: 0

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