PLANET VIP-480FO и trixbox
Здравствуйте!
Имею VIP-480FO с WAN ip-адресом 192.168.99.60 и trixbox с ip-адресом 192.168.99.59
VIP-480FO:
Domain/Realm 192.168.99.59
SIP Proxy Server 192.168.99.59/5060
Hot Line Setting Line 1 167
Line 1 подключен к городской линии
экстеншен 167 на trixbox абсолютно рабочий во всех смыслах
Ситуация:
звоню на номер к которому подключен Line 1 --> ГУДОК --> VIP-480FO поднимает трубку и звонит на 167 экстеншен --> экстеншен 167 звонит - в Line 1 ГУДКОВ НЕТ --> поднимаю трубку 167 экстеншена - НА ОБОИХ КОНЦАХ ТИШИНА !!!
звонок терминируется в обоих направлениях ...
SIP debug + Port Status в аттаче
прошивки перепробовал все: 2.8.2L --> 2.8.6 --> 2.8.8 --> 2.9.1 результат одинаковый
может что не так делаю?
уже всю голову сломал - подскажите?
SIP debug:
<--- SIP read from 192.168.99.60:5060 --->
INVITE sip:167@192.168.99.59 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.60:5060;rport;branch=z9hG4bKdq1259t54m824m9s3364
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>
Contact: <sip:100@192.168.99.60:5060>
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
MAX-Forwards: 70
Content-Type: application/sdp
Content-Length: 294
v=0
o=100 562150 562150 IN IP4 192.168.99.60
s=RTP Audio
c=IN IP4 192.168.99.60
t=0 0
m=audio 2070 RTP/AVP 18 4 108 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:108 FAX/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 13 lines) ---
Sending to 192.168.99.60 : 5060 (NAT)
Using INVITE request as basis request - 19136@192.168.99.60
Found peer 'PLANET 4'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 108
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.99.60:2070
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found unknown media description format FAX for ID 108
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10d (g723|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.99.60:2070
Looking for 167 in from-trunk (domain 192.168.99.59)
list_route: hop: <sip:100@192.168.99.60:5060>
<--- Transmitting (no NAT) to 192.168.99.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.99.60:5060;branch=z9hG4bKdq1259t54m824m9s3364;received=192.168.99.60;rport=5060
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:167@192.168.99.59>
Content-Length: 0
<--- Transmitting (no NAT) to 192.168.99.60:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.99.60:5060;branch=z9hG4bKdq1259t54m824m9s3364;received=192.168.99.60;rport=5060
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:167@192.168.99.59>
Content-Length: 0
<------------>
Audio is at 192.168.99.59 port 19394
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.99.60:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.99.60:5060;branch=z9hG4bKdq1259t54m824m9s3364;received=192.168.99.60;rport=5060
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:167@192.168.99.59>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 3196 3196 IN IP4 192.168.99.59
s=session
c=IN IP4 192.168.99.59
t=0 0
m=audio 19394 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Extension Changed 167[ext-local] new state Ringing for Notify User 155
-- SIP/167-09a322f0 is ringing
-- SIP/167-09a322f0 answered SIP/4000-09a38bd0
Extension Changed 167[ext-local] new state InUse for Notify User 155
Audio is at 192.168.99.59 port 19394
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox_base*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.99.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.99.60:5060;branch=z9hG4bKdq1259t54m824m9s3364;received=192.168.99.60;rport=5060
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:167@192.168.99.59>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 3196 3197 IN IP4 192.168.99.59
s=session
c=IN IP4 192.168.99.59
t=0 0
m=audio 19394 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (no NAT) to 192.168.99.60:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.99.60:5060;branch=z9hG4bKdq1259t54m824m9s3364;received=192.168.99.60;rport=5060
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:167@192.168.99.59>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 3196 3197 IN IP4 192.168.99.59
s=session
c=IN IP4 192.168.99.59
t=0 0
m=audio 19394 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- adaptive jitterbuffer created on channel SIP/4000-09a38bd0
trixbox_base*CLI>
<--- SIP read from 192.168.99.60:5060 --->
ACK sip:167@192.168.99.59 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.60:5060;rport;branch=z9hG4bK1f58uvb77t3p0v5xbes5
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 ACK
MAX-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 192.168.99.60:5060 --->
ACK sip:167@192.168.99.59 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.60:5060;rport;branch=z9hG4bK1f58uvb77t3p0v5xbes5
From: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
To: <sip:167@192.168.99.59>;tag=as191315e3
Call-ID: 19136@192.168.99.60
CSeq: 10489 ACK
MAX-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
trixbox_base*CLI>
<--- SIP read from 192.168.99.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.99.59:5060;branch=z9hG4bK743c6fcd;rport
From: <sip:167@192.168.99.59>;tag=as191315e3
To: <sip:100@192.168.99.59>;tag=teu6i7z824op3d40b553
Call-ID: 19136@192.168.99.60
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '19136@192.168.99.60' Method: ACK
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