Попытка и ACK UDP SIP-пакетов игнорируются при передаче вызова (pjsip) - PullRequest
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Попытка и ACK UDP SIP-пакетов игнорируются при передаче вызова (pjsip)

0 голосов
/ 23 октября 2018

Я хочу перевести вызов из одного контролируемого AGI контекста в другую конечную точку.

То есть.У меня есть вызов, инициированный шлюзом Cisco, для моей конфигурации звездочки, которая обрабатывает некоторую древовидную логику с AGI и в конечном итоге передает вызов другой конечной точке (vbox) 

cisco(62.15.164.62)->
   asterisk(62.12.240.121)-> do some AGI then transfer to vbox
      vbox(62.15.164.65)

Проблема заключается в том, когда происходит передача.Каким-то образом пакеты TRYING и OK SIP из vbox игнорируются звездочкой, так как они не отображаются в CLI, но явно принимаются при просмотре tcpdump udp.И, в конце концов, после того, как моя звездочка повторно выдает пакеты приглашения, вызов сбрасывается, как будто не было получено ни одной попытки / ok, хотя фактически она была отправлена ​​и получена на уровне интерфейса.

Asterisk version 15.5.0

pjsip.conf 

[t-udp-m]
type=transport
protocol=udp
bind=62.12.240.121:5060

[gw1]
type=endpoint
transport=t-udp-m
context=civr
disallow=all
allow=ulaw,speex,gsm
aors=gw1

[gw1]
type=identify
endpoint=gw1
match=62.15.164.62

[gw1]
type=aor
max_contacts=2
remove_existing=yes

[vbox]
type=endpoint
transport=t-udp-m
context=civr
disallow=all
allow=ulaw,alaw,gsm,speex
aors=vbox
send_rpid=yes

[vbox]
type=identify
endpoint=vbox
match=62.15.164.65

[vbox]
type=aor
contact=sip:7011@62.15.164.65:5060
max_contacts=1
remove_existing=yes

extensions.conf 

[civr]
exten => 7010,1,Answer()
 same => n,Agi(/var/www/agi-bin/agi.php)
 same => n,Hangup()

[call_center_altitude]
exten => 1,1,Dial(PJSIP/vbox)
 same => n,Hangup()

tcpdump 

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 1159
INVITE sip:7010@62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8B17BE
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 1380744107-3583906280-2679701539-3944536960
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:0971110799@62.15.164.62>;party=calling;screen=yes;privacy=off
Timestamp: 1540260059
Contact: <sip:0971110799@62.15.164.62:5060>
Call-Info: <sip:62.15.164.62:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1066 6200 IN IP4 62.15.164.62
s=SIP Call
c=IN IP4 62.15.164.62
t=0 0
m=audio 16658 RTP/AVP 0 101
c=IN IP4 62.15.164.62
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 326
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Content-Length:  0


IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 818
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Contact: <sip:62.12.240.121:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   227

v=0
o=- 1066 6202 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 26188 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 371
ACK sip:62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8CF50
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

!!!! The problem starts here.....

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 564
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Length: 0


IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 859
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>;tag=as1dce0520
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 140358089 140358089 IN IP4 62.15.164.65
s=Altitude vBox
c=IN IP4 62.15.164.65
t=0 0
m=audio 25364 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

!!!! As you can see, the last 2 packets were ignored by my asterisk configuration and so it re-issued the INVITE packets...

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

, что в конечном итоге привело к прекращению вызова.

CLI Output 

<--- Received SIP request (1159 bytes) from UDP:62.15.164.62:61396 --->
<INVITE data same as in tcpdump>
  == Setting global variable 'SIPDOMAIN' to '62.12.240.121'
<--- Transmitting SIP response (326 bytes) to UDP:62.15.164.62:61396 --->
<TRYING data same as in tcpdump>
    -- Executing [7010@civr:1] Answer("PJSIP/gw1-00000022", "") in new stack
       > 0x7f707c019640 -- Strict RTP learning after remote address set to: 62.15.164.62:16658
<--- Transmitting SIP response (818 bytes) to UDP:62.15.164.62:61396 --->
<OK data same as in tcpdump>
<--- Received SIP request (371 bytes) from UDP:62.15.164.62:61396 --->
<ACK data same as in tcpdump>
       > 0x7f707c019640 -- Strict RTP switching to RTP target address 62.15.164.62:16658 as source
    -- Executing [7010@civr:8] AGI("PJSIP/gw1-00000022", "/var/www/agi-bin/agi.php") in new stack
    -- Launched AGI Script /var/www/agi-bin/agi.php
<PJSIP/gw1-00000022>AGI Tx >> agi_request: /var/www/agi-bin/agi.php
<PJSIP/gw1-00000022>AGI Tx >> agi_channel: PJSIP/gw1-00000022
<PJSIP/gw1-00000022>AGI Tx >> agi_language: en
<PJSIP/gw1-00000022>AGI Tx >> agi_type: PJSIP
<PJSIP/gw1-00000022>AGI Tx >> agi_uniqueid: 1540257639.63
<PJSIP/gw1-00000022>AGI Tx >> agi_version: 15.5.0
<PJSIP/gw1-00000022>AGI Tx >> agi_callerid: 0971110799
<PJSIP/gw1-00000022>AGI Tx >> agi_calleridname: unknown
<PJSIP/gw1-00000022>AGI Tx >> agi_callingpres: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callingani2: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callington: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callingtns: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_dnid: 7010
<PJSIP/gw1-00000022>AGI Tx >> agi_rdnis: unknown
<PJSIP/gw1-00000022>AGI Tx >> agi_context: civr
<PJSIP/gw1-00000022>AGI Tx >> agi_extension: 7010
<PJSIP/gw1-00000022>AGI Tx >> agi_priority: 8
<PJSIP/gw1-00000022>AGI Tx >> agi_enhanced: 0.0
<PJSIP/gw1-00000022>AGI Tx >> agi_accountcode:
<PJSIP/gw1-00000022>AGI Tx >> agi_threadid: 140122966906624
<PJSIP/gw1-00000022>AGI Tx >>
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-yVABhE ""
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-yVABhE.slin' (escape_digits=) (sample_offset 0) (language 'en')
       > 0x7f707c019640 -- Strict RTP learning complete - Locking on source address 62.15.164.62:16658
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0 endpos=47554
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-9pPsxq "1234"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-9pPsxq.slin' (escape_digits=1234) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=50 endpos=21920
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-zztyOj "12"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-zztyOj.slin' (escape_digits=12) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=49 endpos=12320
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-FyUq0g "#*0123456789"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-FyUq0g.slin' (escape_digits=#*0123456789) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=56 endpos=13440
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=53
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=53
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET CONTEXT call_center_altitude
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET CALLERID 00008577751000971110799
<PJSIP/gw1-00000022>AGI Tx >> 200 result=1
<PJSIP/gw1-00000022>AGI Rx << SET EXTENSION 1
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET PRIORITY 1
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-e3ELZu ""
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-e3ELZu.slin' (escape_digits=) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0 endpos=34697
    -- <PJSIP/gw1-00000022>AGI Script /var/www/agi-bin/agi.php completed, returning 0
    -- Executing [1@call_center_altitude:1] Dial("PJSIP/gw1-00000022", "PJSIP/vbox") in new stack
    -- Called PJSIP/vbox
<--- Transmitting SIP request (1068 bytes) to UDP:62.15.164.65:5060 --->
<INVITE same as in tcpdump>

!!!TRYING и OK SIP-пакеты не отображаются в CLI, как показано в tcpdump !!!

В моей конфигурации конечной точки asterisk vbox может быть что-то не так.

Команда tcpdump: tcpdump -nqt -s 0 -A -i eth0 port 5060, где eth0 - звездочка, не получающая попытку / подтверждение на CLI. 62.12.240.121

Мой ... Я заплачу вам, если вы поможете мне здесь.

1 Ответ

0 голосов
/ 23 октября 2018

Я неправильно прочитал журнал tcpdump.Это показывает, что происходит NAT. 

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 564

Таким образом, моя конфигурация звездочки получила пакет SIP через порт 58140 вместо 5060

...