История изменений [назад]

Ошибки при исходящем вызове

добрый день.

asterisk*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
rt/78827             (Unspecified)                               Auto (No)  No          A  0        Unmonitored
asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
84.53.192.235:5060                      N      78827         285 Registered           Thu, 18 Aug 2016 21:00:49

входящая работает связь, но звук на астериск не идет. Исходящая связь сразу обрывается

[Aug 18 19:48:04] WARNING[2010][C-00000004]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

Дампы прилагаю исходящего. Почему то исходящая начинается с PUBLISH а не с INVITE, С чем ещё необходимо разобраться мне? ..239.114 адрес астериска моего.C:\fakepath\Снимок.JPG

Ошибки при исходящем вызове

добрый день.

asterisk*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
rt/78827             (Unspecified)                               Auto (No)  No          A  0        Unmonitored
asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
84.53.192.235:5060                      N      78827         285 Registered           Thu, 18 Aug 2016 21:00:49

входящая работает связь, но звук на астериск не идет. Исходящая связь сразу обрывается

[Aug 18 19:48:04] WARNING[2010][C-00000004]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

может ещё что-нибудь надо для полноты картины?

Ошибки при исходящем вызове

добрый день.

asterisk*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
rt/78827             (Unspecified)                               Auto (No)  No          A  0        Unmonitored
asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
84.53.192.235:5060                      N      78827         285 Registered           Thu, 18 Aug 2016 21:00:49

входящая работает связь, но звук на астериск не идет. Исходящая связь сразу обрывается

[Aug 18 19:48:04] WARNING[2010][C-00000004]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

может ещё что-нибудь надо для полноты картины?

 <--- SIP read from UDP:192.168.0.148:46836 --->
INVITE sip:89046528473@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: <sip:89046528473@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
 v=0
o=Z 0 0 IN IP4 192.168.0.148
s=Z
c=IN IP4 192.168.0.148
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Sending to 192.168.0.148:46836 (no NAT)
Using INVITE request as basis request - 5QbimmrmaLbeaPXM6VcY6A..
Found peer '111' for '111' from 192.168.0.148:46836
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.148:8000
Looking for 89046528473 in dial (domain 192.168.0.69)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
To: <sip:89046528473@192.168.0.69;transport=udp>
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89046528473@192.168.0.69:5060>
Content-Length: 0
 <------------>
 <--- SIP read from UDP:192.168.0.148:46836 --->
PUBLISH sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---3af1e87c70383b66
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=761ac777
Call-ID: 1RbU6UnPUm3z2nODOz9WNA..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 264
 
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:111@192.168.0.69;transport=UDP"> <tuple id="111" &gt;="" <status=""><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---3af1e87c70383b66;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=761ac777
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as55b3dd06
Call-ID: 1RbU6UnPUm3z2nODOz9WNA..
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog '1RbU6UnPUm3z2nODOz9WNA..' Method: PUBLISH
 <--- SIP read from UDP:192.168.0.148:46836 --->
SUBSCRIBE sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---92ed7d78f95aea2c
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=857cbc2e
Call-ID: W1_y6B22kZUuPqiafjQ5VA..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
 <------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Creating new subscription
Sending to 192.168.0.148:46836 (no NAT)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
Found peer '111' for '111' from 192.168.0.148:46836
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---92ed7d78f95aea2c;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=857cbc2e
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as23478890
Call-ID: W1_y6B22kZUuPqiafjQ5VA..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog 'W1_y6B22kZUuPqiafjQ5VA..' Method: SUBSCRIBE
Really destroying SIP dialog '4ad6e83974bf0ec4604cb0c763f636c3@192.168.0.69:5060' Method: INVITE
<--- Reliably Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
To: <sip:89046528473@192.168.0.69;transport=udp>;tag=as0b29abc6
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
 <------------>
 <--- SIP read from UDP:192.168.0.148:46836 --->
ACK sip:89046528473@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db
Max-Forwards: 70
To: <sip:89046528473@192.168.0.69;transport=udp>;tag=as0b29abc6
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 ACK
Content-Length: 0
 <------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5QbimmrmaLbeaPXM6VcY6A..' Method: ACK
 <--- SIP read from UDP:192.168.0.148:46836 --->
PUBLISH sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---1d470a3823d766f7
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=b20cfe14
Call-ID: TBterlhd5DxA9Ja1UqDqnQ..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 258
 
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:111@192.168.0.69;transport=UDP"> <tuple id="111" &gt;="" <status=""><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---1d470a3823d766f7;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=b20cfe14
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as447537ec
Call-ID: TBterlhd5DxA9Ja1UqDqnQ..
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog 'TBterlhd5DxA9Ja1UqDqnQ..' Method: PUBLISH
 <--- SIP read from UDP:192.168.0.148:46836 --->
SUBSCRIBE sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---a01a78f519333032
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6f63c75f
Call-ID: AKaJR5ZFXkUnlmk6epRl1w..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
 <------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Creating new subscription
Sending to 192.168.0.148:46836 (no NAT)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
Found peer '111' for '111' from 192.168.0.148:46836
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---a01a78f519333032;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6f63c75f
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as6a6226a3
Call-ID: AKaJR5ZFXkUnlmk6epRl1w..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Ошибки при исходящем вызове

добрый день.

asterisk*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
rt/78827             (Unspecified)                               Auto (No)  No          A  0        Unmonitored
asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
84.53.192.235:5060                      N      78827         285 Registered           Thu, 18 Aug 2016 21:00:49

входящая работает связь, но звук на астериск не идет. Исходящая связь сразу обрывается

[Aug 18 19:48:04] WARNING[2010][C-00000004]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

может ещё что-нибудь надо для полноты картины?

 <--- SIP read from UDP:192.168.0.148:46836 --->
INVITE sip:89046528473@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: <sip:89046528473@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
 v=0
o=Z 0 0 IN IP4 192.168.0.148
s=Z
c=IN IP4 192.168.0.148
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Sending to 192.168.0.148:46836 (no NAT)
Using INVITE request as basis request - 5QbimmrmaLbeaPXM6VcY6A..
Found peer '111' for '111' from 192.168.0.148:46836
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.148:8000
Looking for 89046528473 in dial (domain 192.168.0.69)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
To: <sip:89046528473@192.168.0.69;transport=udp>
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89046528473@192.168.0.69:5060>
Content-Length: 0
 <------------>
 <--- SIP read from UDP:192.168.0.148:46836 --->
PUBLISH sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---3af1e87c70383b66
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=761ac777
Call-ID: 1RbU6UnPUm3z2nODOz9WNA..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 264
 
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:111@192.168.0.69;transport=UDP"> <tuple id="111" &gt;="" <status=""><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---3af1e87c70383b66;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=761ac777
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as55b3dd06
Call-ID: 1RbU6UnPUm3z2nODOz9WNA..
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog '1RbU6UnPUm3z2nODOz9WNA..' Method: PUBLISH
 <--- SIP read from UDP:192.168.0.148:46836 --->
SUBSCRIBE sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---92ed7d78f95aea2c
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=857cbc2e
Call-ID: W1_y6B22kZUuPqiafjQ5VA..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
 <------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Creating new subscription
Sending to 192.168.0.148:46836 (no NAT)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
Found peer '111' for '111' from 192.168.0.148:46836
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---92ed7d78f95aea2c;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=857cbc2e
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as23478890
Call-ID: W1_y6B22kZUuPqiafjQ5VA..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog 'W1_y6B22kZUuPqiafjQ5VA..' Method: SUBSCRIBE
Really destroying SIP dialog '4ad6e83974bf0ec4604cb0c763f636c3@192.168.0.69:5060' Method: INVITE
<--- Reliably Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
To: <sip:89046528473@192.168.0.69;transport=udp>;tag=as0b29abc6
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 INVITE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
 <------------>
 <--- SIP read from UDP:192.168.0.148:46836 --->
ACK sip:89046528473@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---99527d7248a967db
Max-Forwards: 70
To: <sip:89046528473@192.168.0.69;transport=udp>;tag=as0b29abc6
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6a297d3f
Call-ID: 5QbimmrmaLbeaPXM6VcY6A..
CSeq: 1 ACK
Content-Length: 0
 <------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5QbimmrmaLbeaPXM6VcY6A..' Method: ACK
 <--- SIP read from UDP:192.168.0.148:46836 --->
PUBLISH sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---1d470a3823d766f7
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=b20cfe14
Call-ID: TBterlhd5DxA9Ja1UqDqnQ..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 258
 
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:111@192.168.0.69;transport=UDP"> <tuple id="111" &gt;="" <status=""><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---1d470a3823d766f7;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=b20cfe14
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as447537ec
Call-ID: TBterlhd5DxA9Ja1UqDqnQ..
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
 <------------>
Really destroying SIP dialog 'TBterlhd5DxA9Ja1UqDqnQ..' Method: PUBLISH
 <--- SIP read from UDP:192.168.0.148:46836 --->
SUBSCRIBE sip:111@192.168.0.69;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---a01a78f519333032
Max-Forwards: 70
Contact: <sip:111@192.168.0.148:46836;transport=udp>
To: "111"<sip:111@192.168.0.69;transport=udp>
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6f63c75f
Call-ID: AKaJR5ZFXkUnlmk6epRl1w..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
 <------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.148:46836 (no NAT)
Creating new subscription
Sending to 192.168.0.148:46836 (no NAT)
list_route: hop: <sip:111@192.168.0.148:46836;transport=udp>
Found peer '111' for '111' from 192.168.0.148:46836
 <--- Transmitting (no NAT) to 192.168.0.148:46836 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.148:46836;branch=z9hG4bK-524287-1---a01a78f519333032;received=192.168.0.148
From: "111"<sip:111@192.168.0.69;transport=udp>;tag=6f63c75f
To: "111"<sip:111@192.168.0.69;transport=udp>;tag=as6a6226a3
Call-ID: AKaJR5ZFXkUnlmk6epRl1w..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
  Настройка транка
 
 [rt]
fromdomain=myip
permit=ipprov/255.255.255.255
port=5060
quality=yes
type=friend
username=78827
secret=password
disallow=all
allow=alaw
allow=ulaw
context=dial
nat=yes