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Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным ~~оброзом~~оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 ~~минут~~минут. А у провайдера 2 минуты и они не сознаються все меня мучают что у меня раньше вырубается

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут. А у провайдера 2 минуты и они не сознаються все меня мучают что у меня раньше вырубается

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Вот debag


Asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:(IP-softPhone):5060 --->
INVITE sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Contact: <sip:1001@(ip-softphone):5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: <sip:1001@(мой ip)="">
Content-Length: 540

v=0
o=- 3435243944 1 IN IP4 (IP-softPhone)
s=SIPPER for PhonerLite
c=IN IP4 (IP-softPhone)
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 112 113 114 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3535838251
a=sendrecv
<------------->
--- (16 headers 22 lines) ---
Sending to (IP-softPhone):5060 (no NAT)
Sending to (IP-softPhone):5060 (no NAT)
Using INVITE request as basis request - 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
Found peer '1001' for '1001' from (IP-softPhone):5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 101
Found unknown media description format opus for ID 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found unknown media description format G726-16 for ID 112
Found unknown media description format G726-24 for ID 113
Found unknown media description format G726-40 for ID 114
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (IP-softPhone):5062
Looking for (Номер вызываемого) in office (domain (мой IP))
list_route: hop: <sip:1001@(ip-softphone):5060>

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
    -- Executing [(Номер вызываемого)@office:1] Dial("SIP/1001-0000000e", "SIP/(провайдер)/(Номер вызываемого),130,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 15174
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to (IP провайдера):5060:
INVITE sip:(Номер вызываемого)@(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1529740561 1529740561 IN IP4 (мой IP)
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 (мой IP)
t=0 0
m=audio 15174 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/(провайдер)/(Номер вызываемого)

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 880077759 880077758 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 7950 RTP/AVP 8 101
c=IN IP4 (IP провайдера)
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1420@(ip провайдера):5060="">
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (IP провайдера):7950
    -- SIP/(провайдер)-0000000f is making progress passing it to SIP/1001-0000000e
Audio is at 14364
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Type: application/sdp
Require: timer
Content-Length: 249

v=0
o=root 1105351112 1105351112 IN IP4 (мой IP)
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 (мой IP)
t=0 0
m=audio 14364 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:(IP провайдера):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 880077759 880077758 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 7950 RTP/AVP 8 101
c=IN IP4 (IP провайдера)
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1420@(ip провайдера):5060="">
    -- SIP/(провайдер)-0000000f is ringing

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
    -- SIP/(провайдер)-0000000f is making progress passing it to SIP/1001-0000000e
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK55c4b558
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as5e7726a8
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK55c4b558
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as5e7726a8
To: <sip:(ip провайдера)="">;tag=1c1127438080
Call-ID: 39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=221/19;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1127461548 1127461547 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '7351224671722017123015@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK79e882ae
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as2ef04194
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK79e882ae
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as2ef04194
To: <sip:1001@(ip-softphone):5060>;tag=00de99c063f3e6118aa6faf648e277c4
Call-ID: 3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1726391199
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c1726376344
To: <sip:(ip провайдера)="">
Call-ID: 17263751661722017123115@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1726391199;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c1726376344
To: <sip:(ip провайдера)="">;tag=as0c39d273
Call-ID: 17263751661722017123115@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '17263751661722017123115@(IP провайдера)' in 32000 ms (Method: OPTIONS)

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK5bcac25a
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as569aa1f9
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 0392e95c39242b745c55231d6913a448@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:48:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK5bcac25a
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as569aa1f9
To: <sip:(ip провайдера)="">;tag=1c2118269270
Call-ID: 0392e95c39242b745c55231d6913a448@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=222/18;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 2118282901 2118282899 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '0392e95c39242b745c55231d6913a448@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '17263751661722017123115@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK28cfc71a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as3deb953f
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:48:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK28cfc71a
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as3deb953f
To: <sip:1001@(ip-softphone):5060>;tag=00245de463f3e6118aa6faf648e277c4
Call-ID: 023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac570154888
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c570132209
To: <sip:(ip провайдера)="">
Call-ID: 5701310501722017123215@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac570154888;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c570132209
To: <sip:(ip провайдера)="">;tag=as5d5e15f7
Call-ID: 5701310501722017123215@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5701310501722017123215@(IP провайдера)' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:(IP-softPhone):5060 --->
CANCEL sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 CANCEL
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to (IP-softPhone):5060 (no NAT)

<--- Reliably Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 CANCEL
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
CANCEL sip:(Номер вызываемого)@(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:(IP-softPhone):5060 --->
ACK sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 ACK
Content-Length: 0

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

<--- SIP read from UDP:(IP-softPhone):5060 --->
ACK sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
  == Spawn extension (office, (Номер вызываемого), 1) exited non-zero on 'SIP/1001-0000000e'
Really destroying SIP dialog '8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)' Method: ACK

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 CANCEL
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to (IP провайдера):5060:
ACK sip:1420@(IP провайдера):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' Method: INVITE
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK555e4127
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as27cfcd1d
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 76263c78175895412fabbdf036052ac4@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:49:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK555e4127
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as27cfcd1d
To: <sip:(ip провайдера)="">;tag=1c961216858
Call-ID: 76263c78175895412fabbdf036052ac4@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=223/17;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 504

v=0
o=AudiocodesGW 961230432 961230430 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '76263c78175895412fabbdf036052ac4@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '5701310501722017123215@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK2543bde7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as6847cf2b
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 736466c9418ec5e13fe89903757acf56@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:49:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK2543bde7
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as6847cf2b
To: <sip:1001@(ip-softphone):5060>;tag=006a200864f3e6118aa6faf648e277c4
Call-ID: 736466c9418ec5e13fe89903757acf56@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '736466c9418ec5e13fe89903757acf56@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1561391498
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c1561376698
To: <sip:(ip провайдера)="">
Call-ID: 15613755441722017123315@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1561391498;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c1561376698
To: <sip:(ip провайдера)="">;tag=as644b7969
Call-ID: 15613755441722017123315@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '15613755441722017123315@(IP провайдера)' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK4329fb03
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as7d4b4955
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 50480b6d393391710a87bb1615a53106@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:50:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK4329fb03
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as7d4b4955
To: <sip:(ip провайдера)="">;tag=1c1952040364
Call-ID: 50480b6d393391710a87bb1615a53106@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=224/16;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1952065087 1952065085 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '50480b6d393391710a87bb1615a53106@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '15613755441722017123315@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK410fdd7a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as69cfc2db
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:50:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK410fdd7a
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as69cfc2db
To: <sip:1001@(ip-softphone):5060>;tag=00b0e32b64f3e6118aa6faf648e277c4
Call-ID: 2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac405153524
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c405138647
To: <sip:(ip провайдера)="">
Call-ID: 4051374861722017123415@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac405153524;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c405138647
To: <sip:(ip провайдера)="">;tag=as323ab2be
Call-ID: 4051374861722017123415@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4051374861722017123415@(IP провайдера)' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK63c8c0ff
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as463604d5
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 595ca09274d317453434c86e1cbc5e84@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:51:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK63c8c0ff
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as463604d5
To: <sip:(ip провайдера)="">;tag=1c795387068
Call-ID: 595ca09274d317453434c86e1cbc5e84@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=224/16;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 504

v=0
o=AudiocodesGW 795400603 795400602 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '595ca09274d317453434c86e1cbc5e84@(мой IP):5060' Method: OPTIONS

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут. А у провайдера 2 минуты и они не сознаються все меня мучают что у меня раньше вырубается

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Вот debag


Asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:(IP-softPhone):5060 --->
INVITE sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Contact: <sip:1001@(ip-softphone):5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: <sip:1001@(мой ip)="">
Content-Length: 540

v=0
o=- 3435243944 1 IN IP4 (IP-softPhone)
s=SIPPER for PhonerLite
c=IN IP4 (IP-softPhone)
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 112 113 114 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3535838251
a=sendrecv
<------------->
--- (16 headers 22 lines) ---
Sending to (IP-softPhone):5060 (no NAT)
Sending to (IP-softPhone):5060 (no NAT)
Using INVITE request as basis request - 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
Found peer '1001' for '1001' from (IP-softPhone):5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 101
Found unknown media description format opus for ID 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found unknown media description format G726-16 for ID 112
Found unknown media description format G726-24 for ID 113
Found unknown media description format G726-40 for ID 114
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (IP-softPhone):5062
Looking for (Номер вызываемого) in office (domain (мой IP))
list_route: hop: <sip:1001@(ip-softphone):5060>

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
    -- Executing [(Номер вызываемого)@office:1] Dial("SIP/1001-0000000e", "SIP/(провайдер)/(Номер вызываемого),130,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 15174
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to (IP провайдера):5060:
INVITE sip:(Номер вызываемого)@(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1529740561 1529740561 IN IP4 (мой IP)
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 (мой IP)
t=0 0
m=audio 15174 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/(провайдер)/(Номер вызываемого)

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 880077759 880077758 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 7950 RTP/AVP 8 101
c=IN IP4 (IP провайдера)
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1420@(ip провайдера):5060="">
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (IP провайдера):7950
    -- SIP/(провайдер)-0000000f is making progress passing it to SIP/1001-0000000e
Audio is at 14364
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Type: application/sdp
Require: timer
Content-Length: 249

v=0
o=root 1105351112 1105351112 IN IP4 (мой IP)
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 (мой IP)
t=0 0
m=audio 14364 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:(IP провайдера):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 880077759 880077758 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 7950 RTP/AVP 8 101
c=IN IP4 (IP провайдера)
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1420@(ip провайдера):5060="">
    -- SIP/(провайдер)-0000000f is ringing

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
    -- SIP/(провайдер)-0000000f is making progress passing it to SIP/1001-0000000e
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK55c4b558
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as5e7726a8
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK55c4b558
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as5e7726a8
To: <sip:(ip провайдера)="">;tag=1c1127438080
Call-ID: 39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=221/19;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1127461548 1127461547 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '39c134ec613ec4c658edf75e3a9d67f1@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '7351224671722017123015@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK79e882ae
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as2ef04194
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:47:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK79e882ae
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as2ef04194
To: <sip:1001@(ip-softphone):5060>;tag=00de99c063f3e6118aa6faf648e277c4
Call-ID: 3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3ee73e6715fe25217e5890d31a3e967b@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1726391199
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c1726376344
To: <sip:(ip провайдера)="">
Call-ID: 17263751661722017123115@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1726391199;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c1726376344
To: <sip:(ip провайдера)="">;tag=as0c39d273
Call-ID: 17263751661722017123115@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '17263751661722017123115@(IP провайдера)' in 32000 ms (Method: OPTIONS)

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:(Номер вызываемого)@(мой="" ip):5060="">
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK5bcac25a
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as569aa1f9
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 0392e95c39242b745c55231d6913a448@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:48:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK5bcac25a
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as569aa1f9
To: <sip:(ip провайдера)="">;tag=1c2118269270
Call-ID: 0392e95c39242b745c55231d6913a448@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=222/18;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 2118282901 2118282899 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '0392e95c39242b745c55231d6913a448@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '17263751661722017123115@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK28cfc71a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as3deb953f
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:48:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK28cfc71a
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as3deb953f
To: <sip:1001@(ip-softphone):5060>;tag=00245de463f3e6118aa6faf648e277c4
Call-ID: 023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '023943ee31e462f73ecde5cd2eab9f2c@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac570154888
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c570132209
To: <sip:(ip провайдера)="">
Call-ID: 5701310501722017123215@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac570154888;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c570132209
To: <sip:(ip провайдера)="">;tag=as5d5e15f7
Call-ID: 5701310501722017123215@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5701310501722017123215@(IP провайдера)' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:(IP-softPhone):5060 --->
CANCEL sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 CANCEL
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to (IP-softPhone):5060 (no NAT)

<--- Reliably Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to (IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;received=(IP-softPhone);rport=5060
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 CANCEL
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
CANCEL sip:(Номер вызываемого)@(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:(IP-softPhone):5060 --->
ACK sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 ACK
Content-Length: 0

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

<--- SIP read from UDP:(IP-softPhone):5060 --->
ACK sip:(Номер вызываемого)@(мой IP) SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">;tag=as5158e360
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
  == Spawn extension (office, (Номер вызываемого), 1) exited non-zero on 'SIP/1001-0000000e'
Really destroying SIP dialog '8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)' Method: ACK

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 CANCEL
Contact: <sip:1420@(ip провайдера):5060="">
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to (IP провайдера):5060:
ACK sip:1420@(IP провайдера):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK734c9a53
Max-Forwards: 70
From: "1001" <sip:8(call id)@(мой="" ip)="">;tag=as445be930
To: <sip:(Номер вызываемого)@(ip="" провайдера)="">;tag=1c879960809
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 137f5b52344ade4c237c1ea859ed347b@(мой IP)
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '137f5b52344ade4c237c1ea859ed347b@(мой IP)' Method: INVITE
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK555e4127
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as27cfcd1d
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 76263c78175895412fabbdf036052ac4@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:49:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK555e4127
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as27cfcd1d
To: <sip:(ip провайдера)="">;tag=1c961216858
Call-ID: 76263c78175895412fabbdf036052ac4@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=223/17;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 504

v=0
o=AudiocodesGW 961230432 961230430 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '76263c78175895412fabbdf036052ac4@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '5701310501722017123215@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK2543bde7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as6847cf2b
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 736466c9418ec5e13fe89903757acf56@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:49:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK2543bde7
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as6847cf2b
To: <sip:1001@(ip-softphone):5060>;tag=006a200864f3e6118aa6faf648e277c4
Call-ID: 736466c9418ec5e13fe89903757acf56@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '736466c9418ec5e13fe89903757acf56@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1561391498
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c1561376698
To: <sip:(ip провайдера)="">
Call-ID: 15613755441722017123315@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac1561391498;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c1561376698
To: <sip:(ip провайдера)="">;tag=as644b7969
Call-ID: 15613755441722017123315@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '15613755441722017123315@(IP провайдера)' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK4329fb03
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as7d4b4955
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 50480b6d393391710a87bb1615a53106@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:50:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK4329fb03
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as7d4b4955
To: <sip:(ip провайдера)="">;tag=1c1952040364
Call-ID: 50480b6d393391710a87bb1615a53106@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=224/16;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1952065087 1952065085 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '50480b6d393391710a87bb1615a53106@(мой IP):5060' Method: OPTIONS
Really destroying SIP dialog '15613755441722017123315@(IP провайдера)' Method: OPTIONS
Reliably Transmitting (no NAT) to (IP-softPhone):5060:
OPTIONS sip:1001@(IP-softPhone):5060 SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK410fdd7a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as69cfc2db
To: <sip:1001@(ip-softphone):5060>
Contact: <sip:asterisk@(мой ip):5060="">
Call-ID: 2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:50:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP-softPhone):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK410fdd7a
From: "asterisk" <sip:asterisk@(мой ip)="">;tag=as69cfc2db
To: <sip:1001@(ip-softphone):5060>;tag=00b0e32b64f3e6118aa6faf648e277c4
Call-ID: 2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:1001@(ip-softphone):5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2f25e5f8555746585f6c9d062ae9e748@(мой IP):5060' Method: OPTIONS

<--- SIP read from UDP:(IP провайдера):5060 --->
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac405153524
Max-Forwards: 70
From: <sip:(ip провайдера)="">;tag=1c405138647
To: <sip:(ip провайдера)="">
Call-ID: 4051374861722017123415@(IP провайдера)
CSeq: 1 OPTIONS
Contact: <sip:(ip провайдера):5060="">
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to (IP провайдера):5060 (no NAT)
Looking for s in default (domain (IP провайдера))

<--- Transmitting (no NAT) to (IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (IP провайдера):5060;branch=z9hG4bKac405153524;received=(IP провайдера)
From: <sip:(ip провайдера)="">;tag=1c405138647
To: <sip:(ip провайдера)="">;tag=as323ab2be
Call-ID: 4051374861722017123415@(IP провайдера)
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:(мой ip):5060="">
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4051374861722017123415@(IP провайдера)' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to (IP провайдера):5060:
OPTIONS sip:(IP провайдера) SIP/2.0
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK63c8c0ff
Max-Forwards: 70
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as463604d5
To: <sip:(ip провайдера)="">
Contact: <sip:8(call id)@(мой="" ip):5060="">
Call-ID: 595ca09274d317453434c86e1cbc5e84@(мой IP):5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Fri, 17 Feb 2017 09:51:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:(IP провайдера):5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP (мой IP):5060;branch=z9hG4bK63c8c0ff
From: "asterisk" <sip:8(call id)@(мой="" ip)="">;tag=as463604d5
To: <sip:(ip провайдера)="">;tag=1c795387068
Call-ID: 595ca09274d317453434c86e1cbc5e84@(мой IP):5060
CSeq: 102 OPTIONS
Contact: <sip:(ip провайдера):5060="">;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=224/16;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 504

v=0
o=AudiocodesGW 795400603 795400602 IN IP4 (IP провайдера)
s=Phone-Call
c=IN IP4 (IP провайдера)
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '595ca09274d317453434c86e1cbc5e84@(мой IP):5060' Method: OPTIONS

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут. А у провайдера 2 минуты и они не сознаються все меня мучают что у меня раньше вырубается

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Вот debag


Asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:(IP-softPhone):5060 UDP:IP-Телефона:5060 --->
INVITE sip:(Номер вызываемого)@(мой IP) sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP (IP-softPhone):5060;branch=z9hG4bK8077eeac63f3e6118aa6faf648e277c4;rport
From: "1001" <sip:1001@(мой ip)="">;tag=1900242408
To: <sip:(Номер вызываемого)@(мой="" ip)="">
Call-ID: 8077EEAC-63F3-E611-8AA5-FAF648E277C4@(IP-softPhone)
CSeq: 28 IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
 Contact: <sip:1001@ip-Телефона:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: <sip:1001@ip-моего_сервака>
Content-Length: 538

v=0
o=- 158410516 1 IN IP4 IP-Телефона
s=SIPPER for PhonerLite
c=IN IP4 IP-Телефона
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 112 113 114 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:959544089
a=sendrecv
<------------->
--- (16 headers 22 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)
Using INVITE request as basis request - 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
Found peer '1001' for '1001' from IP-Телефона:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 101
Found unknown media description format opus for ID 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found unknown media description format G726-16 for ID 112
Found unknown media description format G726-24 for ID 113
Found unknown media description format G726-40 for ID 114
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port IP-Телефона:5062
Looking for 84(вызываемого) in office (domain IP-моего_сервака)
list_route: hop: <sip:1001@ip-Телефона:5060>

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
    -- Executing [84(вызываемого)@office:1] Dial("SIP/1001-00000020", "SIP/rostelecom/84(вызываемого),121,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16216
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
INVITE sip:84(вызываемого)@IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 179552127 179552127 IN IP4 IP-моего_сервака
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 IP-моего_сервака
t=0 0
m=audio 16216 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/rostelecom/84(вызываемого)

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 314683873 314683872 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 8260 RTP/AVP 8 101
c=IN IP4 IP-Провайдера
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1400@ip-Провайдера:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port IP-Провайдера:8260
    -- SIP/rostelecom-00000021 is making progress passing it to SIP/1001-00000020
Audio is at 11606
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 249

v=0
o=root 1074194055 1074194055 IN IP4 IP-моего_сервака
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 IP-моего_сервака
t=0 0
m=audio 11606 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

<------------>
       > 0x7f9840033470 -- Probation passed - setting RTP source address to IP-Телефона:5062
       > 0x7f9860013e60 -- Probation passed - setting RTP source address to IP-Провайдера:8260

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 314683873 314683872 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 8260 RTP/AVP 8 101
c=IN IP4 IP-Провайдера
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1400@ip-Провайдера:5060>
    -- SIP/rostelecom-00000021 is ringing

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
    -- SIP/rostelecom-00000021 is making progress passing it to SIP/1001-00000020
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK077d0245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as2f1df5f8
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK077d0245
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as2f1df5f8
To: <sip:1002@10.101.16.115:5060>;tag=00bb0266cbf5e611851d3a89cc529dd5
Call-ID: 3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '653835802022017135718@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 417 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="426666cd", uri="sip:IP-моего_сервака", response="3c63edf7e47b71a9d4b19280279f1989", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as5c8085a5
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 417 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a402be4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a36624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 418 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="0a402be4", uri="sip:IP-моего_сервака", response="746b55b40b476ed60fdd901cb34b40c4", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1eb8246f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as3c02b56d
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a36624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as5c8085a5
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 418 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:14:58 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1eb8246f
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as3c02b56d
To: <sip:1001@ip-Телефона:5060>;tag=007dee71cbf5e6119a37624fa15194ed
Call-ID: 2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Провайдера:5060 --->
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac1057060349
Max-Forwards: 70
From: <sip:ip-Провайдера>;tag=1c1057045497
To: <sip:ip-Провайдера>
Call-ID: 10570443322022017135818@IP-Провайдера
CSeq: 1 OPTIONS
Contact: <sip:ip-Провайдера:5060>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to IP-Провайдера:5060 (no NAT)
Looking for s in default (domain IP-Провайдера)

<--- Transmitting (no NAT) to IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac1057060349;received=IP-Провайдера
From: <sip:ip-Провайдера>;tag=1c1057045497
To: <sip:ip-Провайдера>;tag=as7c574f89
Call-ID: 10570443322022017135818@IP-Провайдера
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:ip-моего_сервака:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '10570443322022017135818@IP-Провайдера' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1c0b659e
Max-Forwards: 70
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as7dd93796
To: <sip:ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: 6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1c0b659e
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as7dd93796
To: <sip:ip-Провайдера>;tag=1c1176286547
Call-ID: 6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:ip-Провайдера:5060>;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=219/21;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1176300108 1176300107 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060' Method: OPTIONS

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK7f05eb41
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as282884fa
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 549e432724f4707673958edf049c1507@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK7f05eb41
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as282884fa
To: <sip:1002@10.101.16.115:5060>;tag=0001c689cbf5e611851d3a89cc529dd5
Call-ID: 549e432724f4707673958edf049c1507@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '549e432724f4707673958edf049c1507@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '10570443322022017135818@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a37624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 419 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="0a402be4", uri="sip:IP-моего_сервака", response="746b55b40b476ed60fdd901cb34b40c4", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a37624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as65f049ab
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 419 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c76c3bb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a38624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 420 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="3c76c3bb", uri="sip:IP-моего_сервака", response="6f0a66a76fe6e66f3468b42f3177af6a", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK53aafac0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as44671c47
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a38624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as65f049ab
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 420 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:15:54 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK53aafac0
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as44671c47
To: <sip:1001@ip-Телефона:5060>;tag=00694f93cbf5e6119a39624fa15194ed
Call-ID: 47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Провайдера:5060 --->
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac2048728907
Max-Forwards: 70
From: <sip:ip-Провайдера>;tag=1c2048699703
To: <sip:ip-Провайдера>
Call-ID: 20486985352022017135918@IP-Провайдера
CSeq: 1 OPTIONS
Contact: <sip:ip-Провайдера:5060>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to IP-Провайдера:5060 (no NAT)
Looking for s in default (domain IP-Провайдера)

<--- Transmitting (no NAT) to IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac2048728907;received=IP-Провайдера
From: <sip:ip-Провайдера>;tag=1c2048699703
To: <sip:ip-Провайдера>;tag=as1dd0b5fb
Call-ID: 20486985352022017135918@IP-Провайдера
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:ip-моего_сервака:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '20486985352022017135918@IP-Провайдера' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK2cf9824a
Max-Forwards: 70
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as0092c887
To: <sip:ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: 46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK2cf9824a
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as0092c887
To: <sip:ip-Провайдера>;tag=1c19626426
Call-ID: 46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:ip-Провайдера:5060>;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=220/20;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 502

v=0
o=AudiocodesGW 19648920 19648918 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
CANCEL sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 CANCEL
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 CANCEL
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
CANCEL sip:84(вызываемого)@IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:IP-Телефона:5060 --->
ACK sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 ACK
Content-Length: 0

<------------->
  == Spawn extension (office, 84(вызываемого), 1) exited non-zero on 'SIP/1001-00000020'
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона' Method: ACK

<--- SIP read from UDP:IP-Телефона:5060 --->
ACK sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 ACK
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 CANCEL
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to IP-Провайдера:5060:
ACK sip:1400@IP-Провайдера:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Really destroying SIP dialog 'My_Call-id@IP-моего_сервака' Method: INVITE
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1176256c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as01b36186
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1176256c
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as01b36186
To: <sip:1002@10.101.16.115:5060>;tag=004789adcbf5e611851d3a89cc529dd5
Call-ID: 4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '20486985352022017135918@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a39624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 421 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="3c76c3bb", uri="sip:IP-моего_сервака", response="6f0a66a76fe6e66f3468b42f3177af6a", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a39624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as21defb7a
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 421 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36c4baed"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a3a624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 422 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="36c4baed", uri="sip:IP-моего_сервака", response="10a5087440920386cae4f2c2bbb1c783", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK42b700bb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as0a1a4284
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a3a624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as21defb7a
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 422 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:16:50 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK42b700bb
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as0a1a4284
To: <sip:1001@ip-Телефона:5060>;tag=0055b0b4cbf5e6119a3b624fa15194ed
Call-ID: 5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060' Method: OPTIONS
Asterisk*CLI> sip set debug off
SIP Debugging Disabled

или как сменить Reg. default duration: 120 secs на более больше

Отбой на вызов при не ответе

Отбой на вызов при не ответе не срабатывает должным оброзом Мне нужно чтоб более двух минут названивал (2:05 минут) А он вырубает на 1:59 минут. А у провайдера 2 минуты и они не сознаються все меня мучают что у меня раньше вырубается

[office]
exten => _1XXX,1,Dial(SIP/${EXTEN},125)
;exten => _[78]XXXXXXXXXX,1,Dial(SIP/${EXTEN},125,tT)
;exten => s-CANCEL,2,Hangup 

exten => _[78]XXXXXXXXXX,1,Set(CALLERID(num)=8****) 
same => n,Dial(SIP/${EXTEN}@Провайдер,125)
same => s-CANCEL,n,Hangup

Вот ~~debag~~дебаг с момента начала и до конца исходящего звонка


Asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:IP-Телефона:5060 --->
INVITE sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Contact: <sip:1001@ip-Телефона:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: <sip:1001@ip-моего_сервака>
Content-Length: 538

v=0
o=- 158410516 1 IN IP4 IP-Телефона
s=SIPPER for PhonerLite
c=IN IP4 IP-Телефона
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 112 113 114 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:959544089
a=sendrecv
<------------->
--- (16 headers 22 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)
Using INVITE request as basis request - 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
Found peer '1001' for '1001' from IP-Телефона:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found RTP audio format 101
Found unknown media description format opus for ID 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found unknown media description format G726-16 for ID 112
Found unknown media description format G726-24 for ID 113
Found unknown media description format G726-40 for ID 114
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port IP-Телефона:5062
Looking for 84(вызываемого) in office (domain IP-моего_сервака)
list_route: hop: <sip:1001@ip-Телефона:5060>

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
    -- Executing [84(вызываемого)@office:1] Dial("SIP/1001-00000020", "SIP/rostelecom/84(вызываемого),121,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16216
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
INVITE sip:84(вызываемого)@IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 179552127 179552127 IN IP4 IP-моего_сервака
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 IP-моего_сервака
t=0 0
m=audio 16216 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/rostelecom/84(вызываемого)

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 314683873 314683872 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 8260 RTP/AVP 8 101
c=IN IP4 IP-Провайдера
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1400@ip-Провайдера:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port IP-Провайдера:8260
    -- SIP/rostelecom-00000021 is making progress passing it to SIP/1001-00000020
Audio is at 11606
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 249

v=0
o=root 1074194055 1074194055 IN IP4 IP-моего_сервака
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 IP-моего_сервака
t=0 0
m=audio 11606 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

<------------>
       > 0x7f9840033470 -- Probation passed - setting RTP source address to IP-Телефона:5062
       > 0x7f9860013e60 -- Probation passed - setting RTP source address to IP-Провайдера:8260

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Type: application/sdp
Content-Length: 260

v=0
o=AudiocodesGW 314683873 314683872 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 8260 RTP/AVP 8 101
c=IN IP4 IP-Провайдера
a=sendrecv
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1400@ip-Провайдера:5060>
    -- SIP/rostelecom-00000021 is ringing

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
    -- SIP/rostelecom-00000021 is making progress passing it to SIP/1001-00000020
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK077d0245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as2f1df5f8
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK077d0245
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as2f1df5f8
To: <sip:1002@10.101.16.115:5060>;tag=00bb0266cbf5e611851d3a89cc529dd5
Call-ID: 3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3b7688d267fc4254131b99c95e50d14c@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '653835802022017135718@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 417 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="426666cd", uri="sip:IP-моего_сервака", response="3c63edf7e47b71a9d4b19280279f1989", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as5c8085a5
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 417 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a402be4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a36624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 418 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="0a402be4", uri="sip:IP-моего_сервака", response="746b55b40b476ed60fdd901cb34b40c4", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1eb8246f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as3c02b56d
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:14:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK007dee71cbf5e6119a36624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as5c8085a5
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 418 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:14:58 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1eb8246f
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as3c02b56d
To: <sip:1001@ip-Телефона:5060>;tag=007dee71cbf5e6119a37624fa15194ed
Call-ID: 2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2a456ef92285fce543aaf8987e4d70e4@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Провайдера:5060 --->
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac1057060349
Max-Forwards: 70
From: <sip:ip-Провайдера>;tag=1c1057045497
To: <sip:ip-Провайдера>
Call-ID: 10570443322022017135818@IP-Провайдера
CSeq: 1 OPTIONS
Contact: <sip:ip-Провайдера:5060>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to IP-Провайдера:5060 (no NAT)
Looking for s in default (domain IP-Провайдера)

<--- Transmitting (no NAT) to IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac1057060349;received=IP-Провайдера
From: <sip:ip-Провайдера>;tag=1c1057045497
To: <sip:ip-Провайдера>;tag=as7c574f89
Call-ID: 10570443322022017135818@IP-Провайдера
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:ip-моего_сервака:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '10570443322022017135818@IP-Провайдера' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1c0b659e
Max-Forwards: 70
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as7dd93796
To: <sip:ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: 6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1c0b659e
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as7dd93796
To: <sip:ip-Провайдера>;tag=1c1176286547
Call-ID: 6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:ip-Провайдера:5060>;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=219/21;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 506

v=0
o=AudiocodesGW 1176300108 1176300107 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '6b1bbb8707d6369a1c663027080eb17a@IP-моего_сервака:5060' Method: OPTIONS

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:84(вызываемого)@ip-моего_сервака:5060>
Content-Length: 0


<------------>
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK7f05eb41
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as282884fa
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 549e432724f4707673958edf049c1507@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK7f05eb41
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as282884fa
To: <sip:1002@10.101.16.115:5060>;tag=0001c689cbf5e611851d3a89cc529dd5
Call-ID: 549e432724f4707673958edf049c1507@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '549e432724f4707673958edf049c1507@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '10570443322022017135818@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a37624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 419 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="0a402be4", uri="sip:IP-моего_сервака", response="746b55b40b476ed60fdd901cb34b40c4", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a37624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as65f049ab
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 419 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c76c3bb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a38624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 420 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="3c76c3bb", uri="sip:IP-моего_сервака", response="6f0a66a76fe6e66f3468b42f3177af6a", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK53aafac0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as44671c47
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:15:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK00694f93cbf5e6119a38624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as65f049ab
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 420 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:15:54 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK53aafac0
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as44671c47
To: <sip:1001@ip-Телефона:5060>;tag=00694f93cbf5e6119a39624fa15194ed
Call-ID: 47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '47afaf224fd22d4469d922a373775934@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Провайдера:5060 --->
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac2048728907
Max-Forwards: 70
From: <sip:ip-Провайдера>;tag=1c2048699703
To: <sip:ip-Провайдера>
Call-ID: 20486985352022017135918@IP-Провайдера
CSeq: 1 OPTIONS
Contact: <sip:ip-Провайдера:5060>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 2000/v.6.60A.279.005
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to IP-Провайдера:5060 (no NAT)
Looking for s in default (domain IP-Провайдера)

<--- Transmitting (no NAT) to IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Провайдера:5060;branch=z9hG4bKac2048728907;received=IP-Провайдера
From: <sip:ip-Провайдера>;tag=1c2048699703
To: <sip:ip-Провайдера>;tag=as1dd0b5fb
Call-ID: 20486985352022017135918@IP-Провайдера
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:ip-моего_сервака:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '20486985352022017135918@IP-Провайдера' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
OPTIONS sip:IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK2cf9824a
Max-Forwards: 70
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as0092c887
To: <sip:ip-Провайдера>
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: 46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK2cf9824a
From: "asterisk" <sip:84-(call-id)@ip-моего_сервака>;tag=as0092c887
To: <sip:ip-Провайдера>;tag=1c19626426
Call-ID: 46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:ip-Провайдера:5060>;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
X-Resources: telchs=220/20;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 502

v=0
o=AudiocodesGW 19648920 19648918 IN IP4 IP-Провайдера
s=Phone-Call
c=IN IP4 IP-Провайдера
t=0 0
m=audio 6000 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 6002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 22 lines) ---
Really destroying SIP dialog '46da7cee416fc1d563e716dd4f32c873@IP-моего_сервака:5060' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
CANCEL sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 CANCEL
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 CANCEL
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to IP-Провайдера:5060:
CANCEL sip:84(вызываемого)@IP-Провайдера SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:IP-Телефона:5060 --->
ACK sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 ACK
Content-Length: 0

<------------->
  == Spawn extension (office, 84(вызываемого), 1) exited non-zero on 'SIP/1001-00000020'
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона' Method: ACK

<--- SIP read from UDP:IP-Телефона:5060 --->
ACK sip:84(вызываемого)@IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0080aa5dcbf5e6119a35624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=2985622997
To: <sip:84(вызываемого)@ip-моего_сервака>;tag=as38ddb03a
Call-ID: 0080AA5D-CBF5-E611-9A34-624FA15194ED@IP-Телефона
CSeq: 416 ACK
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 CANCEL
Contact: <sip:1400@ip-Провайдера:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Content-Length: 0

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

<--- SIP read from UDP:IP-Провайдера:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 2000/v.6.60A.279.005
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to IP-Провайдера:5060:
ACK sip:1400@IP-Провайдера:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK77143fd5
Max-Forwards: 70
From: "1001" <sip:84-(call-id)@ip-моего_сервака>;tag=as132a22b6
To: <sip:84(вызываемого)@ip-Провайдера>;tag=1c314582692
Contact: <sip:84-(call-id)@ip-моего_сервака:5060>
Call-ID: My_Call-id@IP-моего_сервака
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog 'My_Call-id@IP-моего_сервака' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' Method: REGISTER
Really destroying SIP dialog 'My_Call-id@IP-моего_сервака' Method: INVITE
Reliably Transmitting (no NAT) to 10.101.16.115:5060:
OPTIONS sip:1002@10.101.16.115:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1176256c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as01b36186
To: <sip:1002@10.101.16.115:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.101.16.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK1176256c
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as01b36186
To: <sip:1002@10.101.16.115:5060>;tag=004789adcbf5e611851d3a89cc529dd5
Call-ID: 4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1002@10.101.16.115:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4ee867d17b5cb652640b866262a5da64@IP-моего_сервака:5060' Method: OPTIONS
Really destroying SIP dialog '20486985352022017135918@IP-Провайдера' Method: OPTIONS

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a39624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 421 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="3c76c3bb", uri="sip:IP-моего_сервака", response="6f0a66a76fe6e66f3468b42f3177af6a", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Sending to IP-Телефона:5060 (no NAT)

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a39624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as21defb7a
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 421 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36c4baed"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
REGISTER sip:IP-моего_сервака SIP/2.0
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a3a624fa15194ed;rport
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 422 REGISTER
Contact: <sip:1001@ip-Телефона:5060>;+sip.instance="<urn:uuid:0055d054-3ce8-e611-b02b-f3fd2a0f49d3>"
Authorization: Digest username="1001", realm="asterisk", nonce="36c4baed", uri="sip:IP-моего_сервака", response="10a5087440920386cae4f2c2bbb1c783", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 70
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to IP-Телефона:5060 (no NAT)
Reliably Transmitting (no NAT) to IP-Телефона:5060:
OPTIONS sip:1001@IP-Телефона:5060 SIP/2.0
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK42b700bb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as0a1a4284
To: <sip:1001@ip-Телефона:5060>
Contact: <sip:asterisk@ip-моего_сервака:5060>
Call-ID: 5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Mon, 20 Feb 2017 11:16:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-Телефона:5060;branch=z9hG4bK0055b0b4cbf5e6119a3a624fa15194ed;received=IP-Телефона;rport=5060
From: "1001" <sip:1001@ip-моего_сервака>;tag=43823722
To: "1001" <sip:1001@ip-моего_сервака>;tag=as21defb7a
Call-ID: 009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона
CSeq: 422 REGISTER
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 70
Contact: <sip:1001@ip-Телефона:5060>;expires=70
Date: Mon, 20 Feb 2017 11:16:50 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '009BC58B-9EF5-E611-9892-624FA15194ED@IP-Телефона' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:IP-Телефона:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP-моего_сервака:5060;branch=z9hG4bK42b700bb
From: "asterisk" <sip:asterisk@ip-моего_сервака>;tag=as0a1a4284
To: <sip:1001@ip-Телефона:5060>;tag=0055b0b4cbf5e6119a3b624fa15194ed
Call-ID: 5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060
CSeq: 102 OPTIONS
Contact: <sip:1001@ip-Телефона:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5182174d4554493e623b84b21f8ca6ac@IP-моего_сервака:5060' Method: OPTIONS
Asterisk*CLI> sip set debug off
SIP Debugging Disabled

или как сменить Reg. default duration: 120 secs на более больше

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Проект компании "АТС Дизайн"
Asterisk^® и Digium^® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.

тут описано как это сделать