sip-tcp транк между Avaya 8800 и Астериском
Добавлено: 02 апр 2013, 16:31
Здравствуйте господа специалисты! Подскажите пожалуйста в чем проблема.
На станции Авая обнаружил один свободный sip trunk : display capacity - SIP Trunks (included in 'Trunk ports'): 1
Пытаюсь тестово-подключить Астериск, но не получается. Вот конфиг.
[root]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
tcpenable=yes
[authentication]
[6000]
type=friend
…...........
…............
[avaya]
type=peer
host=192.168.0.1XX — ip-адрес AVAYA
dtmfmode=rfc2833
transport=tcp
context=avaya
disallow=all
allow=ulaw
[root]# cat /etc/asterisk/extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
…..........
…..........
[avaya]
exten => _33XX,1,Dial(SIP/${EXTEN}@avaya,20,)
Подскажите пожалуйста в чем может быть проблема!
На сервере Астериск два сетевых интерфейса :
eth0 192.168.0.1YY – для Авая с ip-адресом 192.168.0.1XX
eth1 192.168.102.4 — к работающим SIP-телефонам.
ast-real*CLI> sip set debug on -------При звонке на номер Авая 3311
SIP Debugging enabled
<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 243
v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966
<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4788ee86"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Authorization: Digest username="6004",realm="asterisk",nonce="4788ee86",uri="sip:3311@192.168.102.4",response="bca1a172f10773e6211c115b44a98ffc",algorithm=MD5
Content-Length: 243
v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/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 192.168.2.22:55508
Looking for 3311 in demo1 (domain 192.168.102.4)
<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 192.168.4.3
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
<------------->
Reliably Transmitting (NAT) to 192.168.4.3
OPTIONS sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d SIP/2.0
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.102.4>;tag=as4e846880
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>
Contact: <sip:asterisk@192.168.102.4:5060>
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Tue, 02 Apr 2013 12:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.4.3:11966 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport=5060
Contact: <sip:192.168.4.3:11966>
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>;tag=4a25b087
From: "asterisk"<sip:asterisk@192.168.102.4>;tag=as4e846880
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060' Method: OPTIONS
ast-real*CLI>
На станции Авая обнаружил один свободный sip trunk : display capacity - SIP Trunks (included in 'Trunk ports'): 1
Пытаюсь тестово-подключить Астериск, но не получается. Вот конфиг.
[root]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
tcpenable=yes
[authentication]
[6000]
type=friend
…...........
…............
[avaya]
type=peer
host=192.168.0.1XX — ip-адрес AVAYA
dtmfmode=rfc2833
transport=tcp
context=avaya
disallow=all
allow=ulaw
[root]# cat /etc/asterisk/extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
…..........
…..........
[avaya]
exten => _33XX,1,Dial(SIP/${EXTEN}@avaya,20,)
Подскажите пожалуйста в чем может быть проблема!
На сервере Астериск два сетевых интерфейса :
eth0 192.168.0.1YY – для Авая с ip-адресом 192.168.0.1XX
eth1 192.168.102.4 — к работающим SIP-телефонам.
ast-real*CLI> sip set debug on -------При звонке на номер Авая 3311
SIP Debugging enabled
<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 243
v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966
<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4788ee86"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Authorization: Digest username="6004",realm="asterisk",nonce="4788ee86",uri="sip:3311@192.168.102.4",response="bca1a172f10773e6211c115b44a98ffc",algorithm=MD5
Content-Length: 243
v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/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 192.168.2.22:55508
Looking for 3311 in demo1 (domain 192.168.102.4)
<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 192.168.4.3
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.4.3:11966 --->
<------------->
Reliably Transmitting (NAT) to 192.168.4.3
OPTIONS sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d SIP/2.0
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.102.4>;tag=as4e846880
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>
Contact: <sip:asterisk@192.168.102.4:5060>
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Tue, 02 Apr 2013 12:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.4.3:11966 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport=5060
Contact: <sip:192.168.4.3:11966>
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>;tag=4a25b087
From: "asterisk"<sip:asterisk@192.168.102.4>;tag=as4e846880
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060' Method: OPTIONS
ast-real*CLI>