звонки внутри сети работают, собщения передаются.
нужно записать поступающий на него любой звонок(до 10шт одномоментно) из локальной подсетки в идеале по h.323 но можно и по sip
добился следующего:
- Код: выделить все
ubbuntuaster01*CLI>
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
[2020-07-13 21:13:33] ERROR[5041]: chan_ooh323.c:1972 ooh323_onReceivedSetup: Unacceptable ip 172.30.3.11
<--- SIP read from UDP:172.30.3.11:5060 --->
INVITE sip:10.202.54.16 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.11:5060;branch=z9hG4bK4288753473-3315
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Supported: ms-forking,timer,replaces
From: sip:172.30.3.11;tag=plcm_4288753555-3315;epid=8212180EEF91CG
To: <sip:10.202.54.16>
Call-ID: 4288752960-3315
CSeq: 1 INVITE
Session-Expires: 90
Contact: <sip:(null)@172.30.3.11:5060;transport=udp>;proxy=replace;+sip.instance="<urn:uuid:9d6da7d2-2ffd-527b-b5b6-864cb7e2d5f2>"
User-Agent:Polycom HDX 8000 HD (Release - 3.0.4_ne-20259)
Content-Type: application/sdp
Content-Length: 1219
v=0
o=POLYCOM 1793433016 0 IN IP4 172.30.3.11
s=-
c=IN IP4 172.30.3.11
b=AS:384
t=0 0
m=audio 49268 RTP/AVP 115 102 9 15 0 8 18 101
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:15 G728/8000
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 0-15
a=sendrecv
m=video 49270 RTP/AVP 116 109 110 111 96 34 31
b=TIAS:384000
a=rtpmap:116 vnd.polycom.lpr/9000
a=fmtp:116 V=1;minPP=0;PP=150;RS=52;RP=10;PS=1400
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42801f; max-mbps=216000; max-fs=3600; sar=13
a=rtpmap:110 H264/90000
a=fmtp:110 profile-level-id=42801f; packetization-mode=1; max-mbps=216000; max-fs=3600; sar=13
a=rtpmap:111 H264/90000
a=fmtp:111 profile-level-id=64001f; packetization-mode=1; max-mbps=216000; max-fs=3600; sar=13
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1;F;J;T
a=rtpmap:34 H263/90000
a=fmtp:34 CIF4=2;CIF=1;QCIF=1;SQCIF=1;F
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1;QCIF=1
a=sendrecv
a=rtcp-fb:* ccm fir
m=application 49272 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv
<------------->
--- (14 headers 41 lines) ---
Sending to 172.30.3.11:5060 (NAT)
Sending to 172.30.3.11:5060 (NAT)
Using INVITE request as basis request - 4288752960-3315
No matching peer for '172.30.3.11' from '172.30.3.11:5060'
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Got SDP version 0 and unique parts [POLYCOM 1793433016 IN IP4 172.30.3.11]
Found RTP audio format 115
Found RTP audio format 102
Found RTP audio format 9
Found RTP audio format 15
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G7221 for ID 115
Found audio description format G7221 for ID 102
Found audio description format G722 for ID 9
Found unknown media description format G728 for ID 15
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 116
Found RTP video format 109
Found RTP video format 110
Found RTP video format 111
Found RTP video format 96
Found RTP video format 34
Found RTP video format 31
Found video description format H264 for ID 109
Found video description format H264 for ID 110
Found video description format H264 for ID 111
Found video description format H263-1998 for ID 96
Found video description format H263 for ID 34
Found video description format H261 for ID 31
Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|mpeg4|h263|h263p), peer - audio=(ulaw|alaw|g722|g729|siren7|siren14)/video=(h261|h263|h263p|h264)/text=(nothing), combined - (ulaw|alaw|g722|h264|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f4d04662af0 -- Strict RTP learning after remote address set to: 172.30.3.11:49268
Peer audio RTP is at port 172.30.3.11:49268
> 0x7f4d0585a860 -- Strict RTP learning after remote address set to: 172.30.3.11:49270
Peer video RTP is at port 172.30.3.11:49270
Looking for s in from-sip-external (domain 10.202.54.16)
sip_route_dump: route/path hop: <sip:(null)@172.30.3.11:5060;transport=udp>
<--- Transmitting (NAT) to 172.30.3.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.3.11:5060;branch=z9hG4bK4288753473-3315;received=172.30.3.11;rport=5060
From: sip:172.30.3.11;tag=plcm_4288753555-3315;epid=8212180EEF91CG
To: <sip:10.202.54.16>
Call-ID: 4288752960-3315
CSeq: 1 INVITE
Server: FPBX-15.0.16.60(17.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:s@10.202.54.16:5060>
Content-Length: 0
<------------>
-- Executing [s@from-sip-external:1] GotoIf("SIP/172.30.3.11-0000001c", "1?setlanguage:checkanon") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s@from-sip-external:2] Set("SIP/172.30.3.11-0000001c", "CHANNEL(language)=en") in new stack
-- Executing [s@from-sip-external:3] GotoIf("SIP/172.30.3.11-0000001c", "0?noanonymous") in new stack
-- Executing [s@from-sip-external:4] Goto("SIP/172.30.3.11-0000001c", "from-trunk,,1") in new stack
-- Goto (from-trunk,s,1)
-- Executing [s@from-trunk:1] NoOp("SIP/172.30.3.11-0000001c", "No DID or CID Match") in new stack
-- Executing [s@from-trunk:2] Answer("SIP/172.30.3.11-0000001c", "") in new stack
Audio is at 11712
Video is at 10.202.54.16:10454
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding video codec h264 to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 172.30.3.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.11:5060;branch=z9hG4bK4288753473-3315;received=172.30.3.11;rport=5060
From: sip:172.30.3.11;tag=plcm_4288753555-3315;epid=8212180EEF91CG
To: <sip:10.202.54.16>;tag=as0a460f8c
Call-ID: 4288752960-3315
CSeq: 1 INVITE
Server: FPBX-15.0.16.60(17.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:s@10.202.54.16:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 689
v=0
o=root 678410993 678410993 IN IP4 10.202.54.16
s=Asterisk PBX 17.5.1
c=IN IP4 10.202.54.16
b=CT:384
t=0 0
m=audio 11712 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10454 RTP/AVP 109 34 96
a=rtpmap:109 H264/90000
a=fmtp:109 max-mbps=216000;max-fs=3600;profile-level-id=42801F
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=2;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:96 h263-1998/90000
a=fmtp:96 SQCIF=1;QCIF=1;CIF=1;CIF4=2;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
m=application 0 RTP/AVP 100
<------------>
<--- SIP read from UDP:172.30.3.11:5060 --->
ACK sip:s@10.202.54.16:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.11:5060;branch=z9hG4bK4288796560-3315
Max-Forwards: 70
From: sip:172.30.3.11;tag=plcm_4288753555-3315;epid=8212180EEF91CG
To: <sip:10.202.54.16>;tag=as0a460f8c
Call-ID: 4288752960-3315
CSeq: 1 ACK
Contact: <sip:(null)@172.30.3.11:5060;transport=udp>;proxy=replace;+sip.instance="<urn:uuid:9d6da7d2-2ffd-527b-b5b6-864cb7e2d5f2>"
User-Agent:Polycom HDX 8000 HD (Release - 3.0.4_ne-20259)
Supported: ms-forking
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
> 0x7f4d04662af0 -- Strict RTP switching to RTP target address 172.30.3.11:49268 as source
-- Executing [s@from-trunk:3] Log("SIP/172.30.3.11-0000001c", "WARNING,Friendly Scanner from 172.30.3.11") in new stack
[2020-07-13 21:13:39] WARNING[5042][C-00000016]: Ext. s:3 @ from-trunk: Friendly Scanner from 172.30.3.11
-- Executing [s@from-trunk:4] Wait("SIP/172.30.3.11-0000001c", "2") in new stack
-- Executing [s@from-trunk:5] Playback("SIP/172.30.3.11-0000001c", "ss-noservice") in new stack
-- <SIP/172.30.3.11-0000001c> Playing 'ss-noservice.gsm' (language 'en')
> 0x7f4d04662af0 -- Strict RTP learning complete - Locking on source address 172.30.3.11:49268
> 0x7f4d0585a860 -- Strict RTP switching to RTP target address 172.30.3.11:49270 as source
> 0x7f4d0585a860 -- Strict RTP learning complete - Locking on source address 172.30.3.11:49270
-- Executing [s@from-trunk:6] SayAlpha("SIP/172.30.3.11-0000001c", "") in new stack
-- Executing [s@from-trunk:7] Hangup("SIP/172.30.3.11-0000001c", "") in new stack
== Spawn extension (from-trunk, s, 7) exited non-zero on 'SIP/172.30.3.11-0000001c'
-- Executing [h@from-trunk:1] Macro("SIP/172.30.3.11-0000001c", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/172.30.3.11-0000001c", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/172.30.3.11-0000001c", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("SIP/172.30.3.11-0000001c", " montior file= ") in new stack
-- Executing [s@macro-hangupcall:5] GotoIf("SIP/172.30.3.11-0000001c", "1?skipagi") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("SIP/172.30.3.11-0000001c", "") in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/172.30.3.11-0000001c' in macro 'hangupcall'
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/172.30.3.11-0000001c'
Scheduling destruction of SIP dialog '4288752960-3315' in 32000 ms (Method: ACK)
Reliably Transmitting (NAT) to 172.30.3.11:5060:
BYE sip:(null)@172.30.3.11:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.202.54.16:5060;branch=z9hG4bK06f6a67c;rport
Max-Forwards: 70
From: <sip:10.202.54.16>;tag=as0a460f8c
To: sip:172.30.3.11;tag=plcm_4288753555-3315;epid=8212180EEF91CG
Call-ID: 4288752960-3315
CSeq: 102 BYE
User-Agent: FPBX-15.0.16.60(17.5.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:172.30.3.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.54.16:5060;branch=z9hG4bK06f6a67c;rport
From: sip:10.202.54.16;tag=as0a460f8c
To: <sip:172.30.3.11>;tag=plcm_4288753555-3315;epid=8212180EEF91CG
Call-ID: 4288752960-3315
CSeq: 102 BYE
Contact: <sip:(null)@172.30.3.11:5060;transport=udp>;proxy=replace;+sip.instance="<urn:uuid:9d6da7d2-2ffd-527b-b5b6-864cb7e2d5f2>"
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
User-Agent:Polycom HDX 8000 HD (Release - 3.0.4_ne-20259)
Supported: ms-forking
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4288752960-3315' Method: ACK
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
<--- SIP read from UDP:10.202.10.87:57983 --->
REGISTER sip:10.202.54.16 SIP/2.0
Via: SIP/2.0/UDP 10.202.10.87:57983;rport;branch=z9hG4bKPjf46f4cf8f9e5416cbc26d9b8040aa1d3
Route: <sip:10.202.54.16;lr>
Max-Forwards: 70
From: "artemPC" <sip:0301@vas>;tag=878f29b122e34906b641722dbb8f9a3a
To: "artemPC" <sip:0301@vas>
Call-ID: f2dd1b43a724408aba4f60c7758524ab
CSeq: 32299 REGISTER
User-Agent: MicroSIP/3.19.31
Contact: "artemPC" <sip:0301@10.202.10.87:57983;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 10.202.10.87:57983 (NAT)
Sending to 10.202.10.87:57983 (NAT)
<--- Transmitting (NAT) to 10.202.10.87:57983 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.202.10.87:57983;branch=z9hG4bKPjf46f4cf8f9e5416cbc26d9b8040aa1d3;received=10.202.10.87;rport=57983
From: "artemPC" <sip:0301@vas>;tag=878f29b122e34906b641722dbb8f9a3a
To: "artemPC" <sip:0301@vas>;tag=as5eb33c1a
Call-ID: f2dd1b43a724408aba4f60c7758524ab
CSeq: 32299 REGISTER
Server: FPBX-15.0.16.60(17.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65fb682f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'f2dd1b43a724408aba4f60c7758524ab' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.202.10.87:57983 --->
REGISTER sip:10.202.54.16 SIP/2.0
Via: SIP/2.0/UDP 10.202.10.87:57983;rport;branch=z9hG4bKPjc4219f41336343adb551681f0e007da4
Route: <sip:10.202.54.16;lr>
Max-Forwards: 70
From: "artemPC" <sip:0301@vas>;tag=878f29b122e34906b641722dbb8f9a3a
To: "artemPC" <sip:0301@vas>
Call-ID: f2dd1b43a724408aba4f60c7758524ab
CSeq: 32300 REGISTER
User-Agent: MicroSIP/3.19.31
Contact: "artemPC" <sip:0301@10.202.10.87:57983;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0301", realm="asterisk", nonce="65fb682f", uri="sip:10.202.54.16", response="ec2dd5e824fc6a5488a3d7e74abadfe8", algorithm=MD5
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.202.10.87:57983 (NAT)
Reliably Transmitting (NAT) to 10.202.10.87:57983:
OPTIONS sip:0301@10.202.10.87:57983;ob SIP/2.0
Via: SIP/2.0/UDP 10.202.54.16:5060;branch=z9hG4bK44cfa73f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as39c9a537
To: <sip:0301@10.202.10.87:57983;ob>
Contact: <sip:Unknown@10.202.54.16:5060>
Call-ID: 3b6e07b7783f0d6b073817e632b3f8ff@10.202.54.16:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.60(17.5.1)
Date: Mon, 13 Jul 2020 13:13:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 10.202.10.87:57983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.10.87:57983;branch=z9hG4bKPjc4219f41336343adb551681f0e007da4;received=10.202.10.87;rport=57983
From: "artemPC" <sip:0301@vas>;tag=878f29b122e34906b641722dbb8f9a3a
To: "artemPC" <sip:0301@vas>;tag=as5eb33c1a
Call-ID: f2dd1b43a724408aba4f60c7758524ab
CSeq: 32300 REGISTER
Server: FPBX-15.0.16.60(17.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:0301@10.202.10.87:57983;ob>;expires=300
Date: Mon, 13 Jul 2020 13:13:58 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '6d8072d402a517ad6f1cb583579c2203@10.202.54.16:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 10.202.10.87:57983:
NOTIFY sip:0301@10.202.10.87:57983;ob SIP/2.0
Via: SIP/2.0/UDP 10.202.54.16:5060;branch=z9hG4bK6aec257e;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as5f7aabac
To: <sip:0301@10.202.10.87:57983;ob>
Contact: <sip:Unknown@10.202.54.16:5060>
Call-ID: 6d8072d402a517ad6f1cb583579c2203@10.202.54.16:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-15.0.16.60(17.5.1)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 87
Messages-Waiting: no
Message-Account: sip:*97@10.202.54.16
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'f2dd1b43a724408aba4f60c7758524ab' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.202.10.87:57983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.54.16:5060;rport=5060;received=10.202.54.16;branch=z9hG4bK44cfa73f
Call-ID: 3b6e07b7783f0d6b073817e632b3f8ff@10.202.54.16:5060
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as39c9a537
To: <sip:0301@10.202.10.87;ob>;tag=z9hG4bK44cfa73f
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.31
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '3b6e07b7783f0d6b073817e632b3f8ff@10.202.54.16:5060' Method: OPTIONS
<--- SIP read from UDP:10.202.10.87:57983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.54.16:5060;rport=5060;received=10.202.54.16;branch=z9hG4bK6aec257e
Call-ID: 6d8072d402a517ad6f1cb583579c2203@10.202.54.16:5060
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as5f7aabac
To: <sip:0301@10.202.10.87;ob>;tag=z9hG4bK6aec257e
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6d8072d402a517ad6f1cb583579c2203@10.202.54.16:5060' Method: NOTIFY
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
Really destroying SIP dialog 'f2dd1b43a724408aba4f60c7758524ab' Method: REGISTER
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
Reliably Transmitting (NAT) to 10.202.10.87:57983:
OPTIONS sip:0301@10.202.10.87:57983;ob SIP/2.0
Via: SIP/2.0/UDP 10.202.54.16:5060;branch=z9hG4bK39cb7c9e;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as30d5191e
To: <sip:0301@10.202.10.87:57983;ob>
Contact: <sip:Unknown@10.202.54.16:5060>
Call-ID: 799d9e432ac8e0e710e05d6547b60a3e@10.202.54.16:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.60(17.5.1)
Date: Mon, 13 Jul 2020 13:14:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.202.10.87:57983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.202.54.16:5060;rport=5060;received=10.202.54.16;branch=z9hG4bK39cb7c9e
Call-ID: 799d9e432ac8e0e710e05d6547b60a3e@10.202.54.16:5060
From: "Unknown" <sip:Unknown@10.202.54.16>;tag=as30d5191e
To: <sip:0301@10.202.10.87;ob>;tag=z9hG4bK39cb7c9e
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.31
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '799d9e432ac8e0e710e05d6547b60a3e@10.202.54.16:5060' Method: OPTIONS
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
<--- SIP read from UDP:10.202.10.87:57983 --->
<------------->
я только разбираюсь и поставил астериск пару дней назад....
собственно вопросы, правильно ли что при попытки прозвона оборудования оно пишется как INVITE sip:10.202.54.16 SIP/2.0 т.е. у него нет номера например там 1114@10.202.54.16
потому что при регистрации телефона у него есть номер для звонка, а тут оборудование звонит напрямую. нужно ли будет каждое оборудование регистрировать на астериске?
я правиильно понимаю что они не смогли договориться какой кодек использовать? Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
пытался настроить Call Recorder но понял что он заточен только для записи файлов mp3\wav и т.п..
подскажите в какою сторону капать? не могу статью найти...
