Вобщем закомментировал в sip_registrations_custom.com все строки регистрации,бомбить перестало.Но теперь исходящих чето нету.
Вот настройки транка:
[idphone](!)
type=friend
host=sip.telecom.kz
fromdomain=sip.telecom.kz
dtmfmode=rfc2833
context=from_idphone
disallow=all
allow=alaw
nat=no
qualify=no
canreinvite=no
insecure=invite
[123456789](idphone)
username=111111
secret=222222
fromuser=111111
<--- SIP read from UDP:192.168.100.44:59759 --->
INVITE sip:363636@192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-d430fa73b461a775-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>
To: <sip:363636@192.168.100.115:5060>
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 410
v=0
o=3cxVCE 68477715 189396990 IN IP4 192.168.100.44
s=3cxVCE Audio Call
c=IN IP4 192.168.100.44
t=0 0
m=audio 40032 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40030 RTP/AVP 34
c=IN IP4 192.168.100.44
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 192.168.100.44:59759 (NAT)
Using INVITE request as basis request - NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
Found peer '20000' for '20000' from 192.168.100.44:59759
<--- Reliably Transmitting (NAT) to 192.168.100.44:59759 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-d430fa73b461a775-1---d8754z-;received=192.168.100.44;rport=59759
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
To: <sip:363636@192.168.100.115:5060>;tag=as3cfe6caa
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="684f6699"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.100.44:59759 --->
ACK sip:363636@192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-d430fa73b461a775-1---d8754z-;rport
Max-Forwards: 70
To: <sip:363636@192.168.100.115:5060>;tag=as3cfe6caa
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.100.44:59759 --->
INVITE sip:363636@192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-5548ed276d3d5217-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>
To: <sip:363636@192.168.100.115:5060>
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Authorization: Digest username="20000",realm="asterisk",nonce="684f6699",uri="sip:363636@192.168.100.115:5060",response="16a16845107d3a9f4bd1be339b247843",algorithm=MD5
Content-Length: 410
v=0
o=3cxVCE 68477715 189396990 IN IP4 192.168.100.44
s=3cxVCE Audio Call
c=IN IP4 192.168.100.44
t=0 0
m=audio 40032 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40030 RTP/AVP 34
c=IN IP4 192.168.100.44
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.100.44:59759 (NAT)
Using INVITE request as basis request - NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
Found peer '20000' for '20000' from 192.168.100.44:59759
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
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 GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x8000e (gsm|ulaw|alaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.44:40032
Looking for 363636 in phones (domain 192.168.100.115)
list_route: hop: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>
<--- Transmitting (NAT) to 192.168.100.44:59759 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-5548ed276d3d5217-1---d8754z-;received=192.168.100.44;rport=59759
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
To: <sip:363636@192.168.100.115:5060>
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:363636@192.168.100.115:5060>
Content-Length: 0
<------------>
-- Executing [363636@phones:1] NoOp("SIP/20000-00000026", "") in new stack
-- Executing [363636@phones:2] Dial("SIP/20000-00000026", "SIP/123456789/363636") in new stack
== Using SIP RTP CoS mark 5
Audio is at 11560
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/123456789/363636
Retransmitting #1 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.100.44:59759 --->
REGISTER sip:192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-8c58d86694500436-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>
To: "user 20000"<sip:20000@192.168.100.115:5060>
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=4b676455
Call-ID: NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.
CSeq: 87 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Authorization: Digest username="20000",realm="asterisk",nonce="7a8b2162",uri="sip:192.168.100.115:5060",response="0995e0e603905f59013a4032882b59ce",algorithm=MD5
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.100.44:59759 (NAT)
<--- Transmitting (NAT) to 192.168.100.44:59759 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-8c58d86694500436-1---d8754z-;received=192.168.100.44;rport=59759
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=4b676455
To: "user 20000"<sip:20000@192.168.100.115:5060>;tag=as43b6dbed
Call-ID: NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.
CSeq: 87 REGISTER
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2be3e7d4"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.100.44:59759 --->
REGISTER sip:192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-f62cec47ad35177c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>
To: "user 20000"<sip:20000@192.168.100.115:5060>
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=4b676455
Call-ID: NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.
CSeq: 88 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Authorization: Digest username="20000",realm="asterisk",nonce="2be3e7d4",uri="sip:192.168.100.115:5060",response="03ef887bb2f435f8db94ec2df6f1a636",algorithm=MD5
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.100.44:59759 (NAT)
<--- Transmitting (NAT) to 192.168.100.44:59759 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-f62cec47ad35177c-1---d8754z-;received=192.168.100.44;rport=59759
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=4b676455
To: "user 20000"<sip:20000@192.168.100.115:5060>;tag=as43b6dbed
Call-ID: NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.
CSeq: 88 REGISTER
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:20000@192.168.100.44:59759;rinstance=ba3e1d63538195e7>;expires=120
Date: Thu, 22 Nov 2012 09:56:59 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.' in 32000 ms (Method: REGISTER)
Retransmitting #5 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Really destroying SIP dialog '
37d7a762089d27201eaf49ce1cde9aca@sip.telecom.kz' Method: INVITE
Really destroying SIP dialog 'NzkyN2MzNDEzMjJhOTJiYmJhOTEzNjgzZGJlYjEyMWQ.' Method: ACK
<--- SIP read from UDP:192.168.100.44:59759 --->
<------------->
Retransmitting #6 (no NAT) to 92.46.61.21:5060:
INVITE sip:
363636@sip.telecom.kz SIP/2.0
Via: SIP/2.0/UDP 192.168.100.115:5060;branch=z9hG4bK34ee9a11
Max-Forwards: 70
From: "user 20000" <sip:
userX@sip.telecom.kz>;tag=as21b72dd0
To: <sip:
363636@sip.telecom.kz>
Contact: <sip:userX@192.168.100.115:5060>
Call-ID:
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Thu, 22 Nov 2012 09:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 963725860 963725860 IN IP4 192.168.100.115
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.100.115
t=0 0
m=audio 11560 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Scheduling destruction of SIP dialog '
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz' in 32000 ms (Method: INVITE)
-- SIP/123456789-00000027 is circuit-busy
Scheduling destruction of SIP dialog '
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [363636@phones:3] Hangup("SIP/20000-00000026", "") in new stack
== Spawn extension (phones, 363636, 3) exited non-zero on 'SIP/20000-00000026'
Scheduling destruction of SIP dialog 'NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.100.44:59759 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-5548ed276d3d5217-1---d8754z-;received=192.168.100.44;rport=59759
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
To: <sip:363636@192.168.100.115:5060>;tag=as69b3e66e
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog '
16ae64dd1e1f86ce0daed5c27e423750@sip.telecom.kz' Method: INVITE
<--- SIP read from UDP:192.168.100.44:59759 --->
ACK sip:363636@192.168.100.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:59759;branch=z9hG4bK-d8754z-5548ed276d3d5217-1---d8754z-;rport
Max-Forwards: 70
To: <sip:363636@192.168.100.115:5060>;tag=as69b3e66e
From: "user 20000"<sip:20000@192.168.100.115:5060>;tag=2b67306b
Call-ID: NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NmZlZGZkMjgzYTA1NTBiNGMxNzE3ZTA1MTM4NmI1MmQ.' Method: REGISTER
<--- SIP read from UDP:192.168.100.44:59759 --->
<------------->
Really destroying SIP dialog 'NTQxZjllNmQyMThmNWNkNThhMjQ4NmM0Y2VkYmE4ODY.' Method: ACK
<--- SIP read from UDP:192.168.100.44:59759 --->
