shema » 22 окт 2018, 06:00
<--- SIP read from UDP:80.253.20.162:5060 --->
INVITE sip:4999689423@10.170.4.59:5060 SIP/2.0
Via: SIP/2.0/UDP 80.253.20.162:5060;branch=z9hG4bK588488ba;rport
From: <sip:84957274288@80.253.20.162>;tag=as379acfa1
To: "Anonymous" <sip:4999689423@anonymous.invalid>;tag=as38a4e6fe
Contact: <sip:84957274288@80.253.20.162>
Call-ID: 0e96016d7f14684d3acc69a924c86bc0@10.170.4.59:5060
CSeq: 102 INVITE
User-Agent: Linksys/SPA962-6.1.5(a)
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 30262 30264 IN IP4 80.253.20.162
s=session
c=IN IP4 80.253.20.162
t=0 0
m=image 4174 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:238
a=T38FaxMaxDatagram:238
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (13 headers 15 lines) ---
Sending to 80.253.20.162:5060 (no NAT)
== Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog 0e96016d7f14684d3acc69a924c86bc0@10.170.4.59:5060
Capabilities: us - (alaw|ulaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (no NAT) to 80.253.20.162:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 80.253.20.162:5060;branch=z9hG4bK588488ba;received=80.253.20.162;rport=5060
From: <sip:84957274288@80.253.20.162>;tag=as379acfa1
To: "Anonymous" <sip:4999689423@anonymous.invalid>;tag=as38a4e6fe
Call-ID: 0e96016d7f14684d3acc69a924c86bc0@10.170.4.59:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.18.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4999689423@10.170.4.59:5060>
Content-Length: 0
<------------>
<--- Reliably Transmitting (no NAT) to 80.253.20.162:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 80.253.20.162:5060;branch=z9hG4bK588488ba;received=80.253.20.162;rport=5060
From: <sip:84957274288@80.253.20.162>;tag=as379acfa1
To: "Anonymous" <sip:4999689423@anonymous.invalid>;tag=as38a4e6fe
Call-ID: 0e96016d7f14684d3acc69a924c86bc0@10.170.4.59:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.18.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0