Проблема при входящем звонке.
Вот лог исходящего звонка:
=========================Sent==============================
INVITE sip:9333@192.168.1.46 SIP/2.0
CSeq: 1 INVITE
From: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
Via: SIP/2.0/UDP 192.168.1.50:5060
To: <sip:9333@192.168.1.46>
Call-ID: ef258cb38126c875@192.168.1.50
Contact: <sip:501@192.168.1.50>
Max-Forwards: 100
User-Agent: SIP EndPoint for Delphi © KaBoom
Expires: 600
Allow: INVITE, ASK, CANCEL, BYE
Content-Length: 241
Content-Type: application/sdp
v=0
o=anonymous 0 0 IN IP4 192.168.1.50
s=SIP EndPoint for Delphi © KaBoom
c=IN IP4 192.168.1.50
m=audio 10000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
=======================Recieved============================
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060
From: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
To: <sip:9333@192.168.1.46>
Call-ID: ef258cb38126c875@192.168.1.50
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9333@192.168.1.46>
Content-Length: 0
=======================Recieved============================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060
From: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
To: <sip:9333@192.168.1.46>;tag=as47a708b8
Call-ID: ef258cb38126c875@192.168.1.50
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9333@192.168.1.46>
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 2065 2065 IN IP4 192.168.1.46
s=session
c=IN IP4 192.168.1.46
t=0 0
m=audio 15688 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
=========================Sent==============================
ACK sip:9333@192.168.1.46 SIP/2.0
CSeq: 1 ACK
From: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
Via: SIP/2.0/UDP 192.168.1.50:5060
To: <sip:9333@192.168.1.46>
Call-ID: ef258cb38126c875@192.168.1.50
Contact: <sip:501@192.168.1.50>
Max-Forwards: 100
User-Agent: SIP EndPoint for Delphi © KaBoom
Expires: 600
Allow: INVITE, ASK, CANCEL, BYE
Content-Length: 0
[=========================РАЗГОВОР========================]
=======================Recieved============================
BYE sip:501@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK345b3437
From: <sip:9333@192.168.1.46>;tag=as47a708b8
To: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
Contact: <sip:9333@192.168.1.46>
Call-ID: ef258cb38126c875@192.168.1.50
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
=========================Sent==============================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK345b3437
From: <sip:9333@192.168.1.46>;tag=as47a708b8
To: Andrey Kostyushin <sip:501@192.168.1.50>;tag=83562;branch=z9hG4bK93f6977c6
Contact: <sip:9333@192.168.1.46>
Call-ID: ef258cb38126c875@192.168.1.50
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
Здесь все ок. Все работает чудесно.
А вот входящий звонок:
=======================Recieved============================
INVITE sip:501@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK647e8911
From: "Cisco ATA Line 1" <sip:asterisk@192.168.1.46>;tag=as1e89aa5c
To: <sip:501@192.168.1.50>
Contact: <sip:asterisk@192.168.1.46>
Call-ID: 781d51f82a2fef0776762d13012fd608@192.168.1.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Nov 2005 11:54:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 2065 2065 IN IP4 192.168.1.46
s=session
c=IN IP4 192.168.1.46
t=0 0
m=audio 12754 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
=========================Sent==============================
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK647e8911
From: "Cisco ATA Line 1" <sip:asterisk@192.168.1.46>;tag=as1e89aa5c
To: <sip:501@192.168.1.50>
Contact: <sip:asterisk@192.168.1.46>
Call-ID: 781d51f82a2fef0776762d13012fd608@192.168.1.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Nov 2005 11:54:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214
=========================Sent==============================
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK647e8911
From: "Cisco ATA Line 1" <sip:asterisk@192.168.1.46>;tag=as1e89aa5c
To: <sip:501@192.168.1.50>
Contact: <sip:asterisk@192.168.1.46>
Call-ID: 781d51f82a2fef0776762d13012fd608@192.168.1.46
CSeq: 102 INVITE
User-Agent: SIP EndPoint for Delphi © KaBoom
Date: Wed, 09 Nov 2005 11:54:37 GMT
Allow: INVITE, ASK, CANCEL, BYE
Content-Type: application/sdp
Content-Length: 214
=========================Sent==============================
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK647e8911
From: "Cisco ATA Line 1" <sip:asterisk@192.168.1.46>;tag=as1e89aa5c
To: <sip:501@192.168.1.50>
Contact: <sip:asterisk@192.168.1.46>
Call-ID: 781d51f82a2fef0776762d13012fd608@192.168.1.46
CSeq: 102 INVITE
User-Agent: SIP EndPoint for Delphi © KaBoom
Date: Wed, 09 Nov 2005 11:54:37 GMT
Allow: INVITE, ASK, CANCEL, BYE
Content-Type: application/sdp
Content-Length: 241
v=0
o=anonymous 0 0 IN IP4 192.168.1.50
s=SIP EndPoint for Delphi © KaBoom
c=IN IP4 192.168.1.50
m=audio 10000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
И вот после моего "200 Ок" мне астериск должен прислать "АСК". Но он его не присылает, хотя RTP-сессия поднимается.
И вообще потом не реагирует на сообщения, т.е. ни о каком "BYE" речи не идет. Гляньте свежим глазом - что в сообщениях не так? Почему * перестает отвечать?
