Доброго времени суток.
Есть ip телефон, но он почему-то не регистрируется на стороне asterisk.
При включении телефона на сервер asterisk не приходят даже заголовки на регистрацию. Порт использую стандартный 5060, в настройках телефона данные указаны верно.
На стороне asterisk использую PJSIP. В качестве транспорта использую UDP.
Если просто совершить звонок с телефона, то он происходит по след. сценарию:
Но трафика по rtp тоже нет. Как можно победить?
<<< INVITE
>>> 401 Unauthorized
<<< ACK
<<< INVITE
Authentication: DIGEST
>>> TRYING
>>> OK
Цитата:
<--- Received SIP request (782 bytes) from UDP:XXX.XXX.XXX.XXX:5264 --->
INVITE sip:909@domain SIP/2.0
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-596-15d58d-7663f86
Max-Forwards: 70
Supported: replaces
Allow: UPDATE
Contact: <sip:615@192.168.0.74:5060>
Allow-Events: talk, hold, conference
User-Agent: LG-Ericsson LIP-8002 v1.1.10scm_a SN/B40EDCB72017 00403578
Content-Type: application/sdp
Content-Length: 219
v=0
o=LGEIPP 27807 27807 IN IP4 192.168.0.74
s=SIP Call
c=IN IP4 192.168.0.74
t=0 0
m=audio 23002 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
<--- Transmitting SIP response (551 bytes) to UDP:XXX.XXX.XXX.XXX:5264 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.74:5060;rport=5264;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-596-15d58d-7663f86
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>;tag=z9hG4bK-596-15d58d-7663f86
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1547644437/eacd7b5236941e86bd0748de0ea3eeaa",opaque="16c5778225d90515",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.0.1
Content-Length: 0
<--- Received SIP request (481 bytes) from UDP:XXX.XXX.XXX.XXX:5264 --->
ACK sip:909@domain SIP/2.0
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>;tag=z9hG4bK-596-15d58d-7663f86
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-596-15d58d-7663f86
Max-Forwards: 70
User-Agent: LG-Ericsson LIP-8002 v1.1.10scm_a SN/B40EDCB72017 00403578
Contact: <sip:615@192.168.0.74:5060>
Content-Length: 0
<--- Received SIP request (1047 bytes) from UDP:XXX.XXX.XXX.XXX:5264 --->
INVITE sip:909@domain SIP/2.0
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-597-15d605-21aabf0c
Max-Forwards: 70
Supported: replaces
Allow: UPDATE
Contact: <sip:615@192.168.0.74:5060>
Authorization: Digest username="615",realm="asterisk",nonce="1547644437/eacd7b5236941e86bd0748de0ea3eeaa",uri="sip:909@domain",response="8e1f29a0455844d41af0f5d14a8f20a0",algorithm=MD5,cnonce="15d605",opaque="16c5778225d90515",qop=auth,nc=00000001
Allow-Events: talk, hold, conference
User-Agent: LG-Ericsson LIP-8002 v1.1.10scm_a SN/B40EDCB72017 00403578
Content-Type: application/sdp
Content-Length: 219
v=0
o=LGEIPP 27807 27808 IN IP4 192.168.0.74
s=SIP Call
c=IN IP4 192.168.0.74
t=0 0
m=audio 23002 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
== Setting global variable 'SIPDOMAIN' to 'domain'
<--- Transmitting SIP response (368 bytes) to UDP:XXX.XXX.XXX.XXX:5264 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.74:5060;rport=5264;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-597-15d605-21aabf0c
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>
CSeq: 2 INVITE
Server: Asterisk PBX 16.0.1
Content-Length: 0
-- Executing [909@call-out:1] Answer("PJSIP/615-000000c8", "") in new stack
> 0x7faa7002d770 -- Strict RTP learning after remote address set to: 192.168.0.74:23002
<--- Transmitting SIP response (827 bytes) to UDP:XXX.XXX.XXX.XXX:5264 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.74:5060;rport=5264;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-597-15d605-21aabf0c
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>;tag=e69af1c2-fd59-4900-ac8e-ffed516533ff
CSeq: 2 INVITE
Server: Asterisk PBX 16.0.1
Contact: <sip:192.168.8.60:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 195
v=0
o=- 27807 27810 IN IP4 192.168.8.60
s=Asterisk
c=IN IP4 192.168.8.60
t=0 0
m=audio 13170 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP request (487 bytes) from UDP:XXX.XXX.XXX.XXX:5264 --->
ACK sip:192.168.8.60:5060 SIP/2.0
From: <sip:615@domain>;tag=9284d8-4a00a8c0-13c4-50029-596-3072d084-596
To: <sip:909@domain>;tag=e69af1c2-fd59-4900-ac8e-ffed516533ff
Call-ID: e72250-4a00a8c0-13c4-50029-596-7e6e8086-596
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-597-15d687-6cf766a5
Max-Forwards: 70
User-Agent: LG-Ericsson LIP-8002 v1.1.10scm_a SN/B40EDCB72017 00403578
Contact: <sip:615@192.168.0.74:5060>
Content-Length: 0
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