Помогите разобраться, не пойму, почему не открывается RTP канал.
Подключен провайдер МТС, с входящими-исходящими звонками проблем нет (голос есть).
Нет голоса, когда совершаешь исходящие звонки через консоль (ниже привел debug при тестовом звонке): channel originate SIP/MTS/89812345678 application Echo.
Порты SIP (udp/5062) и RTP(udp/10000-20000) проброшены на пограничном маршрутизаторе на ip PBX, SIP ALG (cisco 2911) отключен.
С другими sip провайдерами такой проблемы нет.
SIP.CONF
Код: Выделить всё
[general]
bindport = 5062
bindaddr = 0.0.0.0
externip = 176.222.111.111:5062
localnet = 192.168.0.0/16
localnet=172.16.0.0/16
canreinvite=no
rtptimeout = 60
defaultexpiry=120
registertimeout=30
registerattempts=5
qualify=yes
register => sip_E4760XXXXXX@inside.mts.ru:password:sip_E4760XXXXXX_0001@sz.vpbx.mts.ru:5060
[MTS]
type=friend
host=195.34.37.51
outboundproxy = sz.vpbx.mts.ru
fromdomain=inside.mts.ru
fromuser=sip_E4760XXXXXX
username=sip_E4760XXXXXX
authuser=sip_E4760XXXXXX_0001
secret=password
context=from_mts
dtmfmode=rfc2833
nat=force_rport,comedia
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
insecure=invite
qualify=yes
directmedia = nonatКод: Выделить всё
pbx*CLI> channel originate SIP/MTS/89812345678 application Echo
== Using SIP RTP CoS mark 5
Audio is at 16418
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.34.37.51:5060:
INVITE sip:89812345678@195.34.37.51 SIP/2.0
Via: SIP/2.0/UDP 176.222.111.111:5062;branch=z9hG4bK7629e830;rport
Max-Forwards: 70
From: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
To: <sip:89812345678@195.34.37.51>
Contact: <sip:sip_E4760XXXXXX@176.222.111.111:5062>
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Mon, 28 Dec 2020 08:11:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 1790045937 1790045937 IN IP4 176.222.111.111
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 176.222.111.111
t=0 0
m=audio 16418 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
-- Called MTS/89812345678
<--- SIP read from UDP:195.34.37.51:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 176.222.111.111:5062;received=176.222.111.111;branch=z9hG4bK7629e830;rport=5062
From: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
To: <sip:89812345678@195.34.37.51>
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [SIP/MTS-00000023]
<--- SIP read from UDP:195.34.37.51:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 176.222.111.111:5062;received=176.222.111.111;branch=z9hG4bK7629e830;rport=5062
From: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
To: <sip:89812345678@195.34.37.51>;tag=1317797838-1609143094557
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 102 INVITE
Supported:
Contact: <sip:89812345678@195.34.37.51:5060;transport=udp>
Session: Media
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Length: 309
v=0
o=BroadWorks 640825651 1 IN IP4 195.34.37.52
s=-
c=IN IP4 195.34.37.52
t=0 0
m=audio 16694 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
<------------->
--- (13 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:89812345678@195.34.37.51:5060;transport=udp>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7fe22000de20 -- Strict RTP learning after remote address set to: 195.34.37.52:16694
Peer audio RTP is at port 195.34.37.52:16694
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [SIP/MTS-00000023]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [SIP/MTS-00000023]
-- SIP/MTS-00000023 is making progress
<--- SIP read from UDP:195.34.37.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 176.222.111.111:5062;received=176.222.111.111;branch=z9hG4bK7629e830;rport=5062
From: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
To: <sip:89812345678@195.34.37.51>;tag=1317797838-1609143094557
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 102 INVITE
Supported:
Contact: <sip:89812345678@195.34.37.51:5060;transport=udp>
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp
Content-Type: application/sdp
Content-Length: 309
v=0
o=BroadWorks 640825651 1 IN IP4 195.34.37.52
s=-
c=IN IP4 195.34.37.52
t=0 0
m=audio 16694 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
<------------->
--- (13 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:89812345678@195.34.37.51:5060;transport=udp>
Transmitting (NAT) to 195.34.37.51:5060:
ACK sip:89812345678@195.34.37.51:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 176.222.111.111:5062;branch=z9hG4bK5d981daa;rport
Max-Forwards: 70
From: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
To: <sip:89812345678@195.34.37.51>;tag=1317797838-1609143094557
Contact: <sip:sip_E4760XXXXXX@176.222.111.111:5062>
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Content-Length: 0
---
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [SIP/MTS-00000023]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/MTS-00000023]
-- SIP/MTS-00000023 answered
> Launching Echo() on SIP/MTS-00000023
<--- SIP read from UDP:195.34.37.51:5060 --->
BYE sip:sip_E4760XXXXXX@176.222.111.111:5062 SIP/2.0
Via: SIP/2.0/UDP 195.34.37.51:5060;branch=z9hG4bK629gm010889mj5mr9dk0sdkocun43.1
From: <sip:89812345678@195.34.37.51>;tag=1317797838-1609143094557
To: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 338919369 BYE
Max-Forwards: 69
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 195.34.37.51:5060 (NAT)
Scheduling destruction of SIP dialog '2f464098746abbb13702dd9a710d9de0@inside.mts.ru' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 195.34.37.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.34.37.51:5060;branch=z9hG4bK629gm010889mj5mr9dk0sdkocun43.1;received=195.34.37.51;rport=5060
From: <sip:89812345678@195.34.37.51>;tag=1317797838-1609143094557
To: "Anonymous" <sip:sip_E4760XXXXXX@inside.mts.ru:5062>;tag=as032cfd2b
Call-ID: 2f464098746abbb13702dd9a710d9de0@inside.mts.ru
CSeq: 338919369 BYE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<< [ HANGUP (NULL) ] [SIP/MTS-00000023]
