[РЕШЕНО] CommuniGatePro - NAT - Asterisk нет голоса - Телефония: классическая, IP-телефония (VoIP), NGN сети

a-zazell

Posted July 29, 2014 (edited) · Report post

Здравствуйте, asterisk за NAT подключается к ISP, при входящем звонке (от провайдера на АТС) нет голоса (исходящей пока не занимался). В пакетах от прова в SDP поле медиаданных (m=) прилетают разныые IP (из пула, видимо балансируют). АТС клиента шлет трафик на указанные IP и порт, но в трубке тишина. Странная ситуация в консоли asterisk, после вопроизведения ролика со стороны клиента, поступает еще один звонок на exten "h":

Пир, контекст ISP:

register=AUTHNAME:SECRET@AAA.BBB.CCC.249/AUTHNAME

[AUTHNAME]
type=peer
secret=SECRET
qualify=yes
insecure=invite,port
host=AAA.BBB.CCC.249
fromdomain=ISP.DOMAIN.HERE
dtmfmode=rfc2833
defaultuser=AUTHNAME
context=from-trunk-sip-ISP

# asterisk -rx "sip show peers"
Name/username             Host                                    Dyn Forcerport ACL Port     Status
AUTHNAME/AUTHNAME               AAA.BBB.CCC.249                                             5060     OK (61 ms)
# asterisk -rx "sip show registry"
Host                                    dnsmgr Username       Refresh State                Reg.Time
AAA.BBB.CCC.249:5060             N      AUTHNAME             3585 Registered           Tue, 29 Jul 2014 20:20:47
1 SIP registrations.

[from-trunk-sip-ISP]
exten => _.,1,NoOp(***** Call from ${CALLERID(all)} to ${EXTEN} from ISP *****)
exten => _.,n,Answer
exten => _.,n,Playback(en/hello-world)
exten => _.,n,Hangup

Звонок в CLI Asterisk:

localhost*CLI>

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

-- Executing [AUTHNAME@from-trunk-sip-ISP:1] NoOp("SIP/AUTHNAME-000000d2", "***** Call from "+7<CALLER-ID-NUM>" <+7CALLER-ID-NUM> to AUTHNAME from ISP *****") in new stack

-- Executing [AUTHNAME@from-trunk-sip-ISP:2] Answer("SIP/AUTHNAME-000000d2", "") in new stack

-- Executing [AUTHNAME@from-trunk-sip-ISP:3] Playback("SIP/AUTHNAME-000000d2", "en/hello-world") in new stack

Sent RTP packet to AAA.BBB.CCC.130:48410 (type 00, seq 057591, ts 000160, len 000160)

-- <SIP/AUTHNAME-000000d2> Playing 'en/hello-world.ulaw' (language 'en')

Sent RTP packet to AAA.BBB.CCC.130:48410 (type 00, seq 057592, ts 000320, len 000160)

...

Sent RTP packet to AAA.BBB.CCC.130:48410 (type 00, seq 057665, ts 012000, len 000160)

-- Executing [AUTHNAME@from-trunk-sip-ISP:4] Hangup("SIP/AUTHNAME-000000d2", "") in new stack

== Spawn extension (from-trunk-sip-ISP, AUTHNAME, 4) exited non-zero on 'SIP/AUTHNAME-000000d2'

-- Executing [h@from-trunk-sip-ISP:1] NoOp("SIP/AUTHNAME-000000d2", "***** Call from "+7<CALLER-ID-NUM>" <+7CALLER-ID-NUM> to h from ISP *****") in new stack

-- Executing [h@from-trunk-sip-ISP:2] Answer("SIP/AUTHNAME-000000d2", "") in new stack

== Spawn extension (from-trunk-sip-ISP, h, 2) exited non-zero on 'SIP/AUTHNAME-000000d2'

localhost*CLI>

TCPDump соединения:

[root@localhost user]# tcpdump -vpni eth0 net AAA.BBB.CCC.0/16

tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes

20:29:09.449538 IP (tos 0xa0, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 1252)

AAA.BBB.CCC.249.sip > 192.168.1.233.sip: SIP, length: 1224

INVITE sip:AUTHNAME@192.168.1.233:5060 SIP/2.0

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229666-aoncfje;cgp=example.com;rport

Record-Route: <sip:AAA.BBB.CCC.249:5060;lr>

Record-Route: <sip:10.69.96.2:5060;lr>

Record-Route: <sip:rev.16636103-10.69.96.2.dialog.cgatepro;lr>

Max-Forwards: 67

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

Contact: <sip:signode-306020-DAD2A83E_aoncfje-C9F2@AAA.BBB.CCC.249:5060>

CSeq: 1 INVITE

Supported: 100rel,timer,replaces,histinfo,precondition

Session-Expires: 7200

Min-SE: 900

User-Agent: CommuniGatePro-callLeg/5.4.11

Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER

Content-Type: application/sdp

Content-Length: 337

v=0

o=CGPLeg306020 3585286394 1792643198 IN IP4 AAA.BBB.CCC.130

s=-

c=IN IP4 AAA.BBB.CCC.130

t=0 0

m=audio 48408 RTP/AVP 0 8 18 101

c=IN IP4 AAA.BBB.CCC.130

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16,32-49,64-89

a=sendrecv

a=rtcpping:M:3004196:1094723409

20:29:09.450765 IP (tos 0x60, ttl 64, id 33897, offset 0, flags [none], proto UDP (17), length 810)

192.168.1.233.sip > AAA.BBB.CCC.249.sip: SIP, length: 782

SIP/2.0 100 Trying

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229666-aoncfje;cgp=example.com;received=AAA.BBB.CCC.249;rport=5060

Record-Route: <sip:AAA.BBB.CCC.249:5060;lr>

Record-Route: <sip:10.69.96.2:5060;lr>

Record-Route: <sip:rev.16636103-10.69.96.2.dialog.cgatepro;lr>

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 1 INVITE

Server: FPBX-2.11.0(1.8.28.2)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:AUTHNAME@CLIENT.HOST:5060>

Content-Length: 0

20:29:09.459129 IP (tos 0x60, ttl 64, id 33898, offset 0, flags [none], proto UDP (17), length 1133)

192.168.1.233.sip > AAA.BBB.CCC.249.sip: SIP, length: 1105

SIP/2.0 200 OK

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229666-aoncfje;cgp=example.com;received=AAA.BBB.CCC.249;rport=5060

Record-Route: <sip:AAA.BBB.CCC.249:5060;lr>

Record-Route: <sip:10.69.96.2:5060;lr>

Record-Route: <sip:rev.16636103-10.69.96.2.dialog.cgatepro;lr>

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 1 INVITE

Server: FPBX-2.11.0(1.8.28.2)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:AUTHNAME@CLIENT.HOST:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 263

v=0

o=root 660998376 660998376 IN IP4 CLIENT.HOST

s=Asterisk PBX 1.8.28.2

c=IN IP4 CLIENT.HOST

t=0 0

m=audio 11760 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

20:29:09.514295 IP (tos 0xa0, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 571)

AAA.BBB.CCC.249.sip > 192.168.1.233.sip: SIP, length: 543

ACK sip:AUTHNAME@192.168.1.233:5060 SIP/2.0

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229668-aoncfje;cgp=example.com;rport

Max-Forwards: 20

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

Contact: <sip:signode-306020-DAD2A83E_aoncfje-C9F2@AAA.BBB.CCC.249:5060>

CSeq: 1 ACK

User-Agent: CommuniGatePro-callLeg/5.4.11

Content-Length: 0

20:29:09.520304 IP (tos 0xa0, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 1129)

AAA.BBB.CCC.249.sip > 192.168.1.233.sip: SIP, length: 1101

INVITE sip:AUTHNAME@192.168.1.233:5060;maddr=CLIENT.HOST SIP/2.0

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229670-aoncfje;cgp=example.com;rport

Max-Forwards: 20

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

Contact: <sip:signode-306020-DAD2A83E_aoncfje-C9F2@AAA.BBB.CCC.249:5060>

CSeq: 2 INVITE

Supported: 100rel,timer,replaces,histinfo,precondition

Session-Expires: 1800;refresher=uas

Min-SE: 900

User-Agent: CommuniGatePro-callLeg/5.4.11

Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER

Content-Type: application/sdp

Content-Length: 312

v=0

o=CGPLeg306020 3585286394 1792643199 IN IP4 AAA.BBB.CCC.130

s=-

c=IN IP4 AAA.BBB.CCC.130

t=0 0

m=audio 48410 RTP/AVP 0 8 101

c=IN IP4 AAA.BBB.CCC.130

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16,32-49,64-89

a=sendrecv

a=rtcpping:M1:3004197:1868662530

20:29:09.520613 IP (tos 0x60, ttl 64, id 33899, offset 0, flags [none], proto UDP (17), length 677)

192.168.1.233.sip > AAA.BBB.CCC.249.sip: SIP, length: 649

SIP/2.0 100 Trying

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229670-aoncfje;cgp=example.com;received=AAA.BBB.CCC.249;rport=5060

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 2 INVITE

Server: FPBX-2.11.0(1.8.28.2)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:AUTHNAME@CLIENT.HOST:5060>

Content-Length: 0

20:29:09.520698 IP (tos 0x60, ttl 64, id 33900, offset 0, flags [none], proto UDP (17), length 985)

192.168.1.233.sip > AAA.BBB.CCC.249.sip: SIP, length: 957

SIP/2.0 200 OK

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229670-aoncfje;cgp=example.com;received=AAA.BBB.CCC.249;rport=5060

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 2 INVITE

Server: FPBX-2.11.0(1.8.28.2)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:AUTHNAME@CLIENT.HOST:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 263

v=0

o=root 660998376 660998377 IN IP4 CLIENT.HOST

s=Asterisk PBX 1.8.28.2

c=IN IP4 CLIENT.HOST

t=0 0

m=audio 11760 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

20:29:09.568275 IP (tos 0xa0, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 571)

AAA.BBB.CCC.249.sip > 192.168.1.233.sip: SIP, length: 543

ACK sip:AUTHNAME@192.168.1.233:5060 SIP/2.0

Via: SIP/2.0/UDP AAA.BBB.CCC.249:5060;branch=z9hG4bK229672-aoncfje;cgp=example.com;rport

Max-Forwards: 20

From: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

To: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

Contact: <sip:signode-306020-DAD2A83E_aoncfje-C9F2@AAA.BBB.CCC.249:5060>

CSeq: 2 ACK

User-Agent: CommuniGatePro-callLeg/5.4.11

Content-Length: 0

20:29:09.961288 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 200)

192.168.1.233.11760 > AAA.BBB.CCC.130.48410: UDP, length 172

...

20:29:11.441359 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 200)

192.168.1.233.11760 > AAA.BBB.CCC.130.48410: UDP, length 172

20:29:11.482455 IP (tos 0x60, ttl 64, id 33901, offset 0, flags [none], proto UDP (17), length 685)

192.168.1.233.sip > AAA.BBB.CCC.249.sip: SIP, length: 657

BYE sip:signode-306020-DAD2A83E_aoncfje-C9F2@AAA.BBB.CCC.249:5060 SIP/2.0

Via: SIP/2.0/UDP CLIENT.HOST:5060;branch=z9hG4bK7b5ad5e2;rport

Route: <sip:AAA.BBB.CCC.249:5060;lr>,<sip:10.69.96.2:5060;lr>,<sip:rev.16636103-10.69.96.2.dialog.cgatepro;lr>

Max-Forwards: 70

From: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

To: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 102 BYE

User-Agent: FPBX-2.11.0(1.8.28.2)

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0

20:29:11.533050 IP (tos 0xa0, ttl 51, id 0, offset 0, flags [DF], proto UDP (17), length 425)

AAA.BBB.CCC.249.sip > 192.168.1.233.sip: SIP, length: 397

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.233:5060;branch=z9hG4bK7b5ad5e2;rport=5060

From: <sip:+7<EXTEN>@AAA.BBB.CCC.249>;tag=as7e0ae9ca

To: "+7<CALLER-ID-NUM>" <sip:+7<CALLER-ID-NUM>@AAA.BBB.CCC.249>;tag=F14B7BD2-306020-DAD2A83E_aoncfje-C9F2

Call-ID: 4173575e68bf805020c03c916fa8ec71-mdpg4nu@172.24.23.6:5060.egress

CSeq: 102 BYE

Server: CommuniGatePro/5.4.11

Content-Length: 0

^C

85 packets captured

85 packets received by filter

0 packets dropped by kernel

Легенда:

AAA.BBB.CCC. - полсеть прова
CLIENT.HOST - внешний IP клиента
AUTHNAME - логин
SECRET - пароль
EXTEN - номер клиента
CALLER-ID-NUM - с этого номера тестовые звонки

SIP.CONF:

# cat sip_general_additional.conf | grep 'nat\|rtp\|media\|sdp\|extern\|invite'
rtpkeepalive=0
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
nat=no
externip=CLIENT.HOST
directmedia=no

Второй день долблюсь, не пойму с чьей стороны проблема, от саппорта ответа не поступает на письма.

PS1: Для теста цеплял на Asterisk в другой AS, схема такая-же с регистрацией - все ОК.

PS2: В качестве GW ASUS-RT-N12, никаких port-forward нет. Техник на стороне клиента утверждает, что при настройке аккаунта на VoIP телефоне от D-Link (в тойже 192.168.1/24 висит) голос есть, что больше всего смущает.

PS3: Ситуция точно как описано здесь http://voxlink.ru/kb/asterisk-configuration/asterisk-sdp-in-reinvite-ignoresdpversion/, только с нормальной работой со стороны АТС клиента (направляет RTP трафик на заявленный IP,Port)

Edited July 30, 2014 by a-zazell

Share this post

Link to post

Share on other sites

awsswa

Posted July 30, 2014 · Report post

sip alg но асусе выключите

Share this post

Link to post

Share on other sites

a-zazell

Posted July 30, 2014 · Report post

Обновили ASUS, отключили "SIP Passthrough" - голос есть. Спасибо.

Sign In

a-zazell

Share this post

Link to post

Share on other sites

awsswa

Share this post

Link to post

Share on other sites

a-zazell

Share this post

Link to post

Share on other sites

Join the conversation