PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- Transmitting (NAT) to 192.168.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bKPj56d1c0f841c1438d8f8332023fde0376;received=192.168.2.2;rport=5060
From: <sip:300@10.20.0.250>;tag=4680b574bcec447e9fd3b3530390790c
To: <sip:300@10.20.0.250>;tag=as25387d3d
Call-ID: 48b2448fb2244339941c879914cfb85d
CSeq: 18510 REGISTER
Server: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 1800
Contact: <sip:300@192.168.2.2:5060;ob>;expires=1800
Date: Tue, 15 Dec 2015 13:28:34 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '48b2448fb2244339941c879914cfb85d' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.20.0.250:5060;rport=5060;received=10.20.0.250;branch=z9hG4bK22a5eff8
Call-ID: 5f0c330c082adc045506c0646cc92bfb@10.20.0.250:5060
From: "Unknown" <sip:Unknown@10.20.0.250>;tag=as60e7304c
To: <sip:300@192.168.2.2:5060;ob>;tag=z9hG4bK22a5eff8
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5f0c330c082adc045506c0646cc92bfb@10.20.0.250:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.2.2:5060 --->
<------------->
<--- SIP read from UDP:192.168.2.2:5060 --->
<------------->
Really destroying SIP dialog '48b2448fb2244339941c879914cfb85d' Method: REGISTER
localhost*CLI> sip show peer
peers peer
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
300/300 192.168.2.2 D Yes Yes A 5060 OK (7 ms)
400/400 (Unspecified) D Yes Yes A 0 UNKNOWN
79223268715 (Unspecified) Yes Yes 0 Unmonitored
multifon_2/79223268715 133.233.222.55 Yes Yes 5060 Unmonitored
4 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 1 offline]
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
SDP Session Name: Asterisk PBX 11.20.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: 10.20.0.250:0
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,gsm:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 120 secs
Reg. max duration: 3600 secs
Reg. default duration: 180 secs
Sub. min duration 120 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
[300](С ЭТИМ ВСЕ ПЛОХО ОН ПРОБЛЕМНЫЙ В ДРУГОЙ СЕТИ)
deny=0.0.0.0/0.0.0.0
secret=password42
dtmfmode=inband
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
allow=alaw
allow=ulaw
dial=SIP/300
permit=10.42.0.0/255.255.0.0,192.168.2.2/255.255.255.255,10.20.0.0/255.255.255.0
callerid=300 <300>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[400](С НИМ ВСЕ ХОРОШО)
deny=0.0.0.0/0.0.0.0
secret=password42
dtmfmode=inband
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
allow=alaw
allow=ulaw
dial=SIP/400
permit=0.0.0.0/0.0.0.0
callerid=400 <400>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[79223268715]
type=peer
secret=xyymxxxxx
context=from-trunk
[multifon_2]
disallow=all
defaultuser=7922xxxxxxx
type=peer
secret=xxxxxxxxxx
port=5060
nat=force_rport,comedia
insecure=port,invite
host=xxxxxxxxxxxx
fromuser=792232xxxxxx
fromdomain=multifon.ru
dtmfmode=inband
context=incoming
allow=alaw
allow=ulaw
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bKPj56d1c0f841c1438d8f8332023fde0376;received=192.168.2.2;rport=5060
From: <sip:300@10.20.0.250>;tag=4680b574bcec447e9fd3b3530390790c
To: <sip:300@10.20.0.250>;tag=as25387d3d
Call-ID: 48b2448fb2244339941c879914cfb85d
CSeq: 18510 REGISTER
Server: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 1800
Contact: <sip:300@192.168.2.2:5060;ob>;expires=1800
Date: Tue, 15 Dec 2015 13:28:34 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '48b2448fb2244339941c879914cfb85d' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.20.0.250:5060;rport=5060;received=10.20.0.250;branch=z9hG4bK22a5eff8
Call-ID: 5f0c330c082adc045506c0646cc92bfb@10.20.0.250:5060
From: "Unknown" <sip:Unknown@10.20.0.250>;tag=as60e7304c
To: <sip:300@192.168.2.2:5060;ob>;tag=z9hG4bK22a5eff8
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5f0c330c082adc045506c0646cc92bfb@10.20.0.250:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.2.2:5060 --->
<------------->
<--- SIP read from UDP:192.168.2.2:5060 --->
<------------->
Really destroying SIP dialog '48b2448fb2244339941c879914cfb85d' Method: REGISTER
localhost*CLI> sip show peer
peers peer
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
300/300 192.168.2.2 D Yes Yes A 5060 OK (7 ms)
400/400 (Unspecified) D Yes Yes A 0 UNKNOWN
79223268715 (Unspecified) Yes Yes 0 Unmonitored
multifon_2/79223268715 133.233.222.55 Yes Yes 5060 Unmonitored
4 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 1 offline]
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
SDP Session Name: Asterisk PBX 11.20.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: 10.20.0.250:0
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,gsm:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 120 secs
Reg. max duration: 3600 secs
Reg. default duration: 180 secs
Sub. min duration 120 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
[300](С ЭТИМ ВСЕ ПЛОХО ОН ПРОБЛЕМНЫЙ В ДРУГОЙ СЕТИ)
deny=0.0.0.0/0.0.0.0
secret=password42
dtmfmode=inband
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
allow=alaw
allow=ulaw
dial=SIP/300
permit=10.42.0.0/255.255.0.0,192.168.2.2/255.255.255.255,10.20.0.0/255.255.255.0
callerid=300 <300>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[400](С НИМ ВСЕ ХОРОШО)
deny=0.0.0.0/0.0.0.0
secret=password42
dtmfmode=inband
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
allow=alaw
allow=ulaw
dial=SIP/400
permit=0.0.0.0/0.0.0.0
callerid=400 <400>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[79223268715]
type=peer
secret=xyymxxxxx
context=from-trunk
[multifon_2]
disallow=all
defaultuser=7922xxxxxxx
type=peer
secret=xxxxxxxxxx
port=5060
nat=force_rport,comedia
insecure=port,invite
host=xxxxxxxxxxxx
fromuser=792232xxxxxx
fromdomain=multifon.ru
dtmfmode=inband
context=incoming
allow=alaw
allow=ulaw
