asterisk.ru

<--- SIP read from UDP:xxx.xxx.xxx.xxx:xxxx --->
INVITE sip:8xxxxxxxxx@xxx.xxx.xxx.xxx:xxxx;user=phone SIP/2.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bK2bbb0fd13adbdb00
From: "110" <sip:110@xxx.xxx.xxx.xxx>;tag=8f34d32b-21633
To: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx;user=phone>
Call-ID: 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
CSeq: 1144 INVITE
Contact: <sip:110@xxx.xxx.xxx.xxx:xxxx>
Expires: 90
Max-Forwards: 70
Supported: replaces
User-Agent: dlink 12-35-9923797
Content-Type: application/sdp
Content-Length: 231

v=0
o=110 2128827870 2128827870 IN IP4 xxx.xxx.xxx.xxx
s=Session SDP
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 9034 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
<------------->
--- (14 headers 10 lines) ---
Sending to xxx.xxx.xxx.xxx:xxxx (NAT)
Using INVITE request as basis request - 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
Found peer '110' for '110' from xxx.xxx.xxx.xxx:xxxx

<--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:xxxx --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bK2bbb0fd13adbdb00;received=xxx.xxx.xxx.xxx;rport=xxxx
From: "110" <sip:110@xxx.xxx.xxx.xxx>;tag=8f34d32b-21633
To: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx;user=phone>;tag=as6e372392
Call-ID: 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
CSeq: 1144 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d3efe0c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1BF4-1E24-47021633E1C3E56C1892-170@SipHost' in 32000 ms (Method: INVITE)

[Kasterisk2*CLI>
[0K
<--- SIP read from UDP:xxx.xxx.xxx.xxx:xxxx --->
ACK sip:8xxxxxxxxx@xxx.xxx.xxx.xxx:xxxx;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bK2bbb0fd13adbdb00
From: "110" <sip:110@xxx.xxx.xxx.xxx>;tag=8f34d32b-21633
To: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx;user=phone>;tag=as6e372392
Call-ID: 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
CSeq: 1144 ACK
Max-Forwards: 70
Content-Length: 0

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

[Kasterisk2*CLI>
[0K
<--- SIP read from UDP:xxx.xxx.xxx.xxx:xxxx --->
INVITE sip:8xxxxxxxxx@xxx.xxx.xxx.xxx:xxxx;user=phone SIP/2.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bKd57d870390b8bd8b
From: "110" <sip:110@xxx.xxx.xxx.xxx>;tag=8f34d32b-21633
To: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx;user=phone>
Call-ID: 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
CSeq: 1145 INVITE
Contact: <sip:110@xxx.xxx.xxx.xxx:xxxx>
Expires: 90
Max-Forwards: 70
Authorization: Digest username="110",realm="asterisk",nonce="7d3efe0c",uri="sip:8xxxxxxxxx@xxx.xxx.xxx.xxx:xxxx;user=phone",response="92914e489af89435888bbf045744a07d",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-35-9923797
Content-Type: application/sdp
Content-Length: 231

v=0
o=110 2128827870 2128827870 IN IP4 xxx.xxx.xxx.xxx
s=Session SDP
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 9034 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
<------------->
--- (15 headers 10 lines) ---
Sending to xxx.xxx.xxx.xxx:xxxx (NAT)
Using INVITE request as basis request - 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
Found peer '110' for '110' from xxx.xxx.xxx.xxx:xxxx
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port xxx.xxx.xxx.xxx:9034
Looking for 8xxxxxxxxx in russia (domain xxx.xxx.xxx.xxx)
list_route: hop: <sip:110@xxx.xxx.xxx.xxx:xxxx>

<--- Transmitting (NAT) to xxx.xxx.xxx.xxx:xxxx --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bKd57d870390b8bd8b;received=xxx.xxx.xxx.xxx;rport=xxxx
From: "110" <sip:110@xxx.xxx.xxx.xxx>;tag=8f34d32b-21633
To: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx;user=phone>
Call-ID: 1BF4-1E24-47021633E1C3E56C1892-170@SipHost
CSeq: 1145 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8xxxxxxxxx@xxx.xxx.xxx.xxx:xxxx>
Content-Length: 0