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ответил 2012-05-22 09:23:45 +0400

voznyaa Gravatar voznyaa

<--- SIP read from UDP:172.16.0.6:5060 ---> INVITE sip:390009@192.168.3.2;user=phone SIP/2.0 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Contact: <sip:485485@172.16.0.6:5060> Expires:90 Max-Forwards:70 Supported: replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA Content-Type: application/sdp Content-Length: 229

v=0 o=485485 2206034000 2206034000 IN IP4 172.16.0.6 s=Session SDP c=IN IP4 172.16.0.6 t=0 0 m=audio 9000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (14 headers 11 lines) --- Sending to 172.16.0.6:5060 (NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c" Content-Length: 0

<------------> Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 ---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 ACK Max-Forwards:70 Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:172.16.0.6:5060 ---> INVITE sip:390009@192.168.3.2;user=phone SIP/2.0 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Contact: <sip:485485@172.16.0.6:5060> Expires:90 Max-Forwards:70 Authorization:Digest username="485485",realm="asterisk",nonce="7be06c7c",uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5 Supported: replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA Content-Type: application/sdp Content-Length: 229

v=0 o=485485 2206034000 2206034000 IN IP4 172.16.0.6 s=Session SDP c=IN IP4 172.16.0.6 t=0 0 m=audio 9000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (15 headers 11 lines) --- Sending to 172.16.0.6:5060 (NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.16.0.6:9000 Looking for 390009 in local-phones (domain 192.168.3.2) list_route: hop: <sip:485485@172.16.0.6:5060>

<--- Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:390009@192.168.3.2:5060> Content-Length: 0

<------------>

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 Content-Length: 0

<------------>

<--- SIP read from UDP:172.16.0.6:5060 ---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 ACK Max-Forwards:70 Content-Length: 0

<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' Method: ACK

<--- SIP read from UDP:172.16.0.6:5060 ---> INVITE sip:390009@192.168.3.2;user=phone SIP/2.0 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Contact: <sip:485485@172.16.0.6:5060> Expires:90 Max-Forwards:70 Supported: replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA Content-Type: application/sdp Content-Length: 229

v=0 o=485485 2206034000 2206034000 IN IP4 172.16.0.6 s=Session SDP c=IN IP4 172.16.0.6 t=0 0 m=audio 9000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (14 headers 11 lines) --- Sending to 172.16.0.6:5060 (NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c" Content-Length: 0

<------------> Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 ---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 ACK Max-Forwards:70 Content-Length: 0

<-------------> --- (8 headers 0 lines) ---

<--- SIP read from UDP:172.16.0.6:5060 ---> INVITE sip:390009@192.168.3.2;user=phone SIP/2.0 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Contact: <sip:485485@172.16.0.6:5060> Expires:90 Max-Forwards:70 Authorization:Digest username="485485",realm="asterisk",nonce="7be06c7c",uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5 Supported: replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA Content-Type: application/sdp Content-Length: 229

v=0 o=485485 2206034000 2206034000 IN IP4 172.16.0.6 s=Session SDP c=IN IP4 172.16.0.6 t=0 0 m=audio 9000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (15 headers 11 lines) --- Sending to 172.16.0.6:5060 (NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.16.0.6:9000 Looking for 390009 in local-phones (domain 192.168.3.2) list_route: hop: <sip:485485@172.16.0.6:5060>

<--- Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone> Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:390009@192.168.3.2:5060> Content-Length: 0

<------------>

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 Content-Length: 0

<------------>

<--- SIP read from UDP:172.16.0.6:5060 ---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 ACK Max-Forwards:70 Content-Length: 0

<-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' Method: ACK


<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229

229


v=0
o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060

172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c"
Content-Length: 0

0


<------------>
Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost'
in 32000 ms (Method: INVITE)

INVITE)

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 ACK
Max-Forwards:70
Content-Length: 0

0

<------------->
--- (8 headers 0 lines) ---

---

<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Authorization:Digest
username="485485",realm="asterisk",nonce="7be06c7c",uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229


v=0
o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.6:9000
Looking for 390009 in local-phones (domain 192.168.3.2)
list_route: hop: <sip:485485@172.16.0.6:5060>


<--- Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:390009@192.168.3.2:5060>
Content-Length: 0


<------------>


<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


<------------>


<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 ACK
Max-Forwards:70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' Method: ACK


<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone> sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Contact: <sip:485485@172.16.0.6:5060> sip:485485@172.16.0.6:5060
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229


v=0
o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost'
in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as44d8f4dc sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 ACK
Max-Forwards:70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone> sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Authorization:Digest
username="485485",realm="asterisk",nonce="7be06c7c",uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229


v=0
o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.6:9000
Looking for 390009 in local-phones (domain 192.168.3.2)
list_route: hop: <sip:485485@172.16.0.6:5060>sip:485485@172.16.0.6:5060


<--- Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone> sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:390009@192.168.3.2:5060> sip:390009@192.168.3.2:5060
Content-Length: 0


<------------>


<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 sip:390009@192.168.3.2;user=phone;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


<------------>


<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: <sip:390009@192.168.3.2;user=phone>;tag=as41249ff2 sip:390009@192.168.3.2;user=phone;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 ACK
Max-Forwards:70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' Method: ACK


Method: ACK

Method: ACK> <--- SIP read from

UDP:172.16.0.6:5060 ---> INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839 To: sip:390009@192.168.3.2;user=phone Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Contact: sip:485485@172.16.0.6:5060
Expires:90 Max-Forwards:70
Supported: replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229 v=0 o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP c=IN IP4 172.16.0.6
t=0 0 m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 a=sendrecv
<-------------> --- (14 headers 11 lines) --- Sending to 172.16.0.6:5060 (NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060 <--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060 From: "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839 To: sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c" Content-Length: 0

<------------> Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' in 32000 ms (Method: INVITE) <--- SIP read from UDP:172.16.0.6:5060 ---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1472 ACK Max-Forwards:70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2 From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839 To: sip:390009@192.168.3.2;user=phone Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Contact: sip:485485@172.16.0.6:5060
Expires:90
Max-Forwards:70
Contact: <sip:485485@172.16.0.6:5060> Expires:90 Max-Forwards:70 Authorization:Digest username="485485", realm="asterisk", nonce="7be06c7c", uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5 Supported: replaces
replaces User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
dlink 12-3868-1709-0.10.56.1-DSA Content-Type: application/sdp
application/sdp Content-Length: 229


v=0
229

v=0 o=485485 2206034000 2206034000 2206034000 IN IP4 172.16.0.6
172.16.0.6 s=Session SDP
c=IN SDP c=IN IP4 172.16.0.6
172.16.0.6 t=0 0
0 m=audio 9000 9000 RTP/AVP 8 101
101 a=rtpmap:8 PCMA/8000
PCMA/8000 a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
vad=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
a=sendrecv
<------------->
0-15 a=sendrecv <-------------> --- (14 (15 headers 11 lines) ---
--- Sending to to 172.16.0.6:5060 (NAT)
(NAT) Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
- BD21-1117-47098839C1AB44C93C22-010@SipHost Found peer '485485' for '485485' from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to from 172.16.0.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060
From: "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be06c7c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost'
in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1472 ACK
Max-Forwards:70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
To: sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
CSeq: 1473 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Authorization:Digest
username="485485",realm="asterisk",nonce="7be06c7c",uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229


v=0
o=485485 2206034000 2206034000 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-47098839C1AB44C93C22-010@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060
Found RTP audio format 8
8 Found RTP audio format 101
101 Found audio description format PCMA PCMA for ID 8
8 Found audio description description format telephone-event for ID 101
101 Capabilities: us - 0x8 (alaw), peer - - audio=0x8 (alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined combined - 0x8 (alaw)
(alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), (telephone-event|), peer - 0x1 (telephone-event|), (telephone-event|), combined - 0x1 (telephone-event|)
(telephone-event|) Peer audio RTP is at port 172.16.0.6:9000
port 172.16.0.6:9000 Looking for 390009 in local-phones (domain 192.168.3.2)
192.168.3.2) list_route: hop: hop: sip:485485@172.16.0.6:5060


<--- Transmitting (NAT) to to 172.16.0.6:5060 --->
---> SIP/2.0 100 Trying
Trying Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: sip:390009@192.168.3.2;user=phone
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
sip:390009@192.168.3.2;user=phone Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE
INVITE Server: Asterisk PBX
Asterisk PBX Allow: INVITE, ACK, CANCEL, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: PUBLISH Supported: replaces, timer
Contact: sip:390009@192.168.3.2:5060
timer Contact: sip:390009@192.168.3.2:5060 Content-Length: 0


<------------>


<------------> <--- Reliably Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
172.16.0.6:5060 ---> SIP/2.0 503 Service Unavailable
Unavailable Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
From: "485485" "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: To: sip:390009@192.168.3.2;user=phone;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 INVITE
INVITE Server: Asterisk PBX
Asterisk PBX Allow: INVITE, ACK, CANCEL, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: PUBLISH Supported: replaces, timer
timer X-Asterisk-HangupCause: User busy
busy X-Asterisk-HangupCauseCode: 17
17 Content-Length: 0


<------------>


<--- SIP read from from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
---> ACK sip:390009@192.168.3.2;user=phone SIP/2.0 Via: SIP/2.0/UDP SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
From: "485485" "485485" sip:485485@192.168.3.2;tag=d72bcd7a-98839
To: To: sip:390009@192.168.3.2;user=phone;tag=as41249ff2
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost
Call-ID: BD21-1117-47098839C1AB44C93C22-010@SipHost CSeq: 1473 ACK
Max-Forwards:70
ACK Max-Forwards:70 Content-Length: 0


<------------->
<-------------> --- (8 headers 0 0 lines) ---
--- Really destroying SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' SIP dialog 'BD21-1117-47098839C1AB44C93C22-010@SipHost' Method: ACKMethod: ACK

Method: ACK

Method: ACK

Method: ACK> 

<--- SIP read from
 
 
from UDP:172.16.0.6:5060 --->   --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 INVITE
  sip:390009@192.168.3.2;user=phone
  SIP/2.0 

  Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
  Via: SIP/2.0/UDP
  172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
  From: "485485"
  sip:485485@192.168.3.2;tag=d72bcd7a-98839
  To: sip:390009@192.168.3.2;user=phone 
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost   CSeq: 1472 INVITE   Contact:
  sip:485485@172.16.0.6:5060 

  Expires:90   Max-Forwards:70 
 Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Supported: replaces   User-Agent:
  replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA 
 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp 
 application/sdp
Content-Length: 229
       v=0   
v=0
o=485485 2206034000 2206034000 2207805700 2207805700 IN IP4 172.16.0.6 
 172.16.0.6
s=Session SDP   SDP
c=IN IP4 172.16.0.6 
 172.16.0.6
t=0 0   0
m=audio 9000 RTP/AVP 8 101 
 101
a=rtpmap:8 PCMA/8000   PCMA/8000
a=fmtp:8 vad=no
 a=rtpmap:101 telephone-event/8000 
 telephone-event/8000
a=fmtp:101 0-15   a=sendrecv 

  <------------->   0-15
a=sendrecv
<------------->
--- (14 headers 11
  11 lines) ---  ---
Sending to 172.16.0.6:5060
  (NAT)  Using INVITE request as basis
  request -
  BD21-1117-47098839C1AB44C93C22-010@SipHost  Found peer '485485' for '485485' from
  172.16.0.6:5060    <--- Reliably Transmitting (NAT) to 172.16.0.6:5060
  --->  SIP/2.0 401 Unauthorized  Via: SIP/2.0/UDP 

  172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc;received=172.16.0.6;rport=5060
  From: "485485"
  sip:485485@192.168.3.2;tag=d72bcd7a-98839
  To:
  sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1472 INVITE  Server: Asterisk
  PBX  Allow: INVITE, ACK, CANCEL,
  OPTIONS, BYE, REFER, SUBSCRIBE,
  NOTIFY, INFO, PUBLISH  Supported:
  replaces, timer  WWW-Authenticate:
  Digest algorithm=MD5,
  realm="asterisk", nonce="7be06c7c" 
  Content-Length: 0
 
<------------>  Scheduling
  destruction of SIP dialog
  'BD21-1117-47098839C1AB44C93C22-010@SipHost'
  in 32000 ms (Method: INVITE)    <---
  SIP read from UDP:172.16.0.6:5060 --->
  ACK sip:390009@192.168.3.2;user=phone
  SIP/2.0  Via: SIP/2.0/UDP
  172.16.0.6:5060;branch=z9hG4bK5ef832975c9d1ddc
  From: "485485"
  <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
  To:
  sip:390009@192.168.3.2;user=phone;tag=as44d8f4dc
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1472 ACK  Max-Forwards:70 
  Content-Length: 0    <-------------> 
  --- (8 headers 0 lines) ---    <--- SIP read from UDP:172.16.0.6:5060 --->
  INVITE
  sip:390009@192.168.3.2;user=phone
  SIP/2.0 

  Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
  Via: SIP/2.0/UDP
  172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
  From: "485485"
  <sip:485485@192.168.3.2>;tag=d72bcd7a-98839
  To: sip:390009@192.168.3.2;user=phone 
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1473 INVITE  Contact:
  <sip:485485@172.16.0.6:5060> 
  Expires:90  Max-Forwards:70 
  Authorization:Digest 
  username="485485", realm="asterisk",
  nonce="7be06c7c",
  uri="sip:390009@192.168.3.2;user=phone",response="cfc6a6eee9d64805cd7b010e8a50df71",algorithm=MD5
  Supported: replaces  User-Agent: dlink
  12-3868-1709-0.10.56.1-DSA 
  Content-Type: application/sdp 
  Content-Length: 229
 
v=0   o=485485 2206034000 2206034000
  IN IP4 172.16.0.6  s=Session SDP  c=IN
  IP4 172.16.0.6  t=0 0  m=audio 9000
  RTP/AVP 8 101  a=rtpmap:8 PCMA/8000 
  a=fmtp:8 vad=no  a=rtpmap:101
  telephone-event/8000  a=fmtp:101 0-15 
  a=sendrecv  <------------->  --- (15
  headers 11 lines) ---  Sending to
  172.16.0.6:5060 (NAT)  (NAT)
Using INVITE request as basis request -
  BD21-1117-47098839C1AB44C93C22-010@SipHost  - BD21-1117-471006102A80D015A4BF-012@SipHost
Found peer '485485' for '485485' from
  from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060  --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as1b333b47
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f62044a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BD21-1117-471006102A80D015A4BF-012@SipHost' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as1b333b47
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 ACK
Max-Forwards:70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Authorization:Digest username="485485",realm="asterisk",nonce="7f62044a",uri="sip:390009@192.168.3.2;user=phone",response="46ad1c02712007c4f7f31706c4443e5b",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229

v=0
o=485485 2207805700 2207805700 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-471006102A80D015A4BF-012@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060
Found RTP audio format 8  8
Found RTP audio format 101 
  101
Found audio description format PCMA
  PCMA for ID 8  8
Found audio description
  description format telephone-event for ID 101 
  101
Capabilities: us - 0x8 (alaw), peer -
  - audio=0x8 (alaw)/video=0x0 
 (nothing)/text=0x0 (nothing), combined
  combined - 0x8 (alaw)  (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|),
  (telephone-event|), peer - 0x1 (telephone-event|),
  (telephone-event|), combined - 0x1 (telephone-event|) 
  (telephone-event|)
Peer audio RTP is at port
  172.16.0.6:9000  port 172.16.0.6:9000
Looking for 390009 in local-phones (domain 192.168.3.2) 
  192.168.3.2)
list_route: hop:
  sip:485485@172.16.0.6:5060
 
hop: <sip:485485@172.16.0.6:5060>

<--- Transmitting (NAT) to
  to 172.16.0.6:5060 --->  --->
SIP/2.0 100 Trying  Via: SIP/2.0/UDP Trying
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:390009@192.168.3.2:5060>
Content-Length: 0


<------------>

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as356451b2
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


<------------>

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as356451b2
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 ACK
Max-Forwards:70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'BD21-1117-471006102A80D015A4BF-012@SipHost' Method: ACK
 172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
  From: "485485"
  sip:485485@192.168.3.2;tag=d72bcd7a-98839
  To: sip:390009@192.168.3.2;user=phone 
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1473 INVITE  Server: Asterisk
  PBX  Allow: INVITE, ACK, CANCEL,
  OPTIONS, BYE, REFER, SUBSCRIBE,
  NOTIFY, INFO, PUBLISH  Supported:
  replaces, timer  Contact:
  sip:390009@192.168.3.2:5060 
  Content-Length: 0
 
<------------>  <--- Reliably
  Transmitting (NAT) to 172.16.0.6:5060
  --->  SIP/2.0 503 Service Unavailable  Via: SIP/2.0/UDP 
   172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2;received=172.16.0.6;rport=5060
  From: "485485"
  sip:485485@192.168.3.2;tag=d72bcd7a-98839
  To:
  sip:390009@192.168.3.2;user=phone;tag=as41249ff2
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1473 INVITE  Server: Asterisk
  PBX  Allow: INVITE, ACK, CANCEL,
  OPTIONS, BYE, REFER, SUBSCRIBE,
  NOTIFY, INFO, PUBLISH  Supported:
  replaces, timer 
  X-Asterisk-HangupCause: User busy 
  X-Asterisk-HangupCauseCode: 17 
  Content-Length: 0
 
<------------>
 
<--- SIP read from
  UDP:172.16.0.6:5060 --->  ACK
  sip:390009@192.168.3.2;user=phone
  SIP/2.0  Via: SIP/2.0/UDP
  172.16.0.6:5060;branch=z9hG4bK7ffd690e0e9180f2
  From: "485485"
  sip:485485@192.168.3.2;tag=d72bcd7a-98839
  To:
  sip:390009@192.168.3.2;user=phone;tag=as41249ff2
  Call-ID:
  BD21-1117-47098839C1AB44C93C22-010@SipHost  CSeq: 1473 ACK  Max-Forwards:70 
  Content-Length: 0
 
<------------->  --- (8 headers 0
  lines) ---  Really destroying SIP
  dialog
  'BD21-1117-47098839C1AB44C93C22-010@SipHost'
  Method: ACKMethod: ACK
 
 
Method: ACK
 enter code here
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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.