asterisk.ru

Все сделал. Приду с обеда - кину логи. Звонок с мобильного на панас не идёт.

После внесения всех изменений слетел FPBX. Работает 5-10 минут и вырубается. Слава богу есть снапшот.
Позже напишу что и как. Надеюсь на дальнейшую помощь.

Господи ... вы еще и на виртуалке все это запускаете ...

awsswa писал(а):ага ... дружно поржали
где, я спрашиваю где хоть в одном мануале сказано что номер может начинаться на ноль ?

у вас стоит - qualify=yes
значит телефон должен отображаться что зарегистрирован - типа - OK (11 ms) - передерните питание на телефоне

Индивидуальное обучение начинающих путём бездумного исполнения ими команд? В которых ни фига не понимают? В конце - Пасиба огромное?
Не мечите бисер, человеку нужно документацию для начала почитать.

правильный пинок на старте - творит чудеса

awsswa писал(а):Включаете все кодаки

приоритет по порядку
alaw 1
ulaw 2
g729 3
g722 4
g726 5

потом проверяете регистрацию и делаете лог звонка как в 1 вопросе

"Соединение невозможно, перезвоните позднее"...

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Дебаг

Код: Выделить всё

89-28-166-183*CLI> sip set debug on
SIP Debugging re-enabled
Really destroying SIP dialog '837687589303395156099559' Method: OPTIONS

<--- SIP read from UDP:89.28.160.23:5061 --->
INVITE sip:69801@89.28.166.183;user=phone SIP/2.0
Via: SIP/2.0/UDP 89.28.160.23:5061;rport;branch=z9hG4bK-2187499719-3792783037-817677987-203481435
From: <sip:9113574005@89.28.160.23:5061;user=phone>;tag=3227228359-3792783037-817677987-203481435
To: <sip:69801@89.28.166.183;user=phone>
Call-ID: c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23
CSeq: 1 INVITE
Contact: <sip:9113574005@89.28.160.23:5061;user=phone>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.4.0-16
Cisco-Guid: 905682723-3162378722-2254869996-81909911
Content-Length: 389

v=0
o=- 1368612503 1368612503 IN IP4 89.28.160.23
s=-
c=IN IP4 89.28.160.23
t=0 0
m=audio 13818 RTP/AVP 8 18 97 98 99 96
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 G729/8000
a=fmtp:97 annexb=no
a=rtpmap:98 G729/8000
a=fmtp:98 annexb=yes
a=rtpmap:99 G729/8000
a=fmtp:99 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 89.28.160.23:5061 (no NAT)
Using INVITE request as basis request - c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23
No matching peer for '9113574005' from '89.28.160.23:5061'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 96
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G729 for ID 97
Found audio description format G729 for ID 98
Found audio description format G729 for ID 99
Found audio description format telephone-event for ID 96
Capabilities: us - 0x80004 (ulaw|h263), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[May 13 16:42:46] NOTICE[9402]: chan_sip.c:8897 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 89.28.160.23:5061 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 89.28.160.23:5061;branch=z9hG4bK-2187499719-3792783037-817677987-203481435;received=89.28.160.23;rport=5061
From: <sip:9113574005@89.28.160.23:5061;user=phone>;tag=3227228359-3792783037-817677987-203481435
To: <sip:69801@89.28.166.183;user=phone>;tag=as1038ab7f
Call-ID: c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:89.28.160.23:5061 --->
ACK sip:69801@89.28.166.183;user=phone SIP/2.0
Via: SIP/2.0/UDP 89.28.160.23:5061;rport;branch=z9hG4bK-2187499719-3792783037-817677987-203481435
From: <sip:9113574005@89.28.160.23:5061;user=phone>;tag=3227228359-3792783037-817677987-203481435
To: <sip:69801@89.28.166.183;user=phone>;tag=as1038ab7f
Call-ID: c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.4.0-16
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'c7985be8bd4611e2a3c6bc305be1200c@89.28.160.23' Method: ACK
Reliably Transmitting (NAT) to 89.28.162.105:28952:
OPTIONS sip:69777@172.22.222.42:5060 SIP/2.0
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK0f8d251f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@89.28.166.183>;tag=as0d38fc6a
To: <sip:69777@172.22.222.42:5060>
Contact: <sip:Unknown@89.28.166.183:5060>
Call-ID: 6f4030252d4ae5ec04fc3ece19f14013@89.28.166.183:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Mon, 13 May 2013 13:42:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:89.28.162.105:28952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK0f8d251f;rport=5060;received=89.28.166.183
Call-ID: 6f4030252d4ae5ec04fc3ece19f14013@89.28.166.183:5060
From: "Unknown" <sip:Unknown@89.28.166.183>;tag=as0d38fc6a
To: <sip:69777@172.22.222.42:5060>;tag=3879262100
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
Contact: <sip:172.22.222.42:5060>
Server: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6f4030252d4ae5ec04fc3ece19f14013@89.28.166.183:5060' Method: OPTIONS

судя по логу - и вас asterisk с провайдером договорится по кодакам не могут, и до панаса дело даже не доходит.
Провайдер дает только g729 и alaw - что у вас выставлено на провайдера ?

добавте в [MAPT]
insecure=port,invite
у вас не мачится на этот пир

awsswa писал(а):судя по логу - и вас asterisk с провайдером договорится по кодакам не могут, и до панаса дело даже не доходит.
Провайдер дает только g729 и alaw - что у вас выставлено на провайдера ?

Провайдер даёт alaw

Vlad1983 писал(а):добавте в [MAPT]
insecure=port,invite
у вас не мачится на этот пир

Добавил. Теперь панас принимает звонки, всё отлично. Но с панаса на мобильный не позвонить. Пробую разбираться. Вот лог:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: ЛОГ

Код: Выделить всё

<--- SIP read from UDP:89.28.162.105:28952 --->
INVITE sip:89113574005@89.28.166.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs4570d571f
Max-Forwards: 70
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
Supported: replaces
Contact: <sip:69777@172.22.222.42:5060>
Content-Type: application/sdp
User-Agent: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 317

v=0
o=- 1368454063 1368454063 IN IP4 172.22.222.42
s=-
c=IN IP4 172.22.222.42
t=0 0
m=audio 16136 RTP/AVP 8 0 18 9 2 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (13 headers 15 lines) ---
Sending to 89.28.162.105:5060 (no NAT)
Using INVITE request as basis request - 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
Found peer '69777' for '69777' from 89.28.162.105:28952

<--- Reliably Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs4570d571f;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as71ac735e
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="138a695a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42' in 9856 ms (Method: INVITE)

<--- SIP read from UDP:89.28.162.105:28952 --->
ACK sip:89113574005@89.28.166.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs4570d571f
Max-Forwards: 70
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as71ac735e
CSeq: 1 ACK
User-Agent: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 0

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

<--- SIP read from UDP:89.28.162.105:28952 --->
INVITE sip:89113574005@89.28.166.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf
Max-Forwards: 70
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>
CSeq: 2 INVITE
Authorization: Digest realm="asterisk",nonce="138a695a",algorithm=MD5,uri="sip:89113574005@89.28.166.183:5060",username="69777",response="0777df44c6d0b1136bf718f58168ec7c"
Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
Supported: replaces
Contact: <sip:69777@172.22.222.42:5060>
Content-Type: application/sdp
User-Agent: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 317

v=0
o=- 1368454063 1368454063 IN IP4 172.22.222.42
s=-
c=IN IP4 172.22.222.42
t=0 0
m=audio 16136 RTP/AVP 8 0 18 9 2 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (14 headers 15 lines) ---
Sending to 89.28.162.105:28952 (NAT)
Using INVITE request as basis request - 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
Found peer '69777' for '69777' from 89.28.162.105:28952
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190c (ulaw|alaw|g726|g729|g722), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x190c (ulaw|alaw|g726|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.222.42:16136
Looking for 89113574005 in from-internal (domain 89.28.166.183:5060)
list_route: hop: <sip:69777@172.22.222.42:5060>

<--- Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:89113574005@89.28.166.183:5060>
Content-Length: 0


<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 89.28.160.23:5060:
INVITE sip:89113574005@89.28.160.23:5060 SIP/2.0
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
Max-Forwards: 70
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>
Contact: <sip:69801@89.28.166.183:5060>
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Mon, 13 May 2013 14:07:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 389

v=0
o=root 535274474 535274474 IN IP4 89.28.166.183
s=Asterisk PBX 1.8.7.1
c=IN IP4 89.28.166.183
t=0 0
m=audio 15364 RTP/AVP 8 0 18 9 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:89.28.160.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 INVITE
Contact: <sip:89113574005@89.28.160.23:5060>
Server: MERA MVTS3G v.4.4.0-16
Content-Length: 0

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

<--- SIP read from UDP:89.28.160.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>;tag=1048899395-3792784061-817677987-203481435
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 INVITE
Contact: <sip:89113574005@89.28.160.23:5060>
Content-Type: application/sdp
Server: MERA MVTS3G v.4.4.0-16
Content-Length: 288

v=0
o=- 1368614000 1368614000 IN IP4 89.28.160.23
s=-
c=IN IP4 89.28.160.23
t=0 0
m=audio 14000 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (10 headers 14 lines) ---
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190c (ulaw|alaw|g726|g729|g722), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 89.28.160.23:14000

<--- Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as3788c253
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:89113574005@89.28.166.183:5060>
Content-Length: 0


<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as3788c253
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:89113574005@89.28.166.183:5060>
Content-Type: application/sdp
Content-Length: 387

v=0
o=root 2051884123 2051884123 IN IP4 89.28.166.183
s=Asterisk PBX 1.8.7.1
c=IN IP4 89.28.166.183
t=0 0
m=audio 19326 RTP/AVP 8 0 18 9 2 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:89.28.162.105:28952 --->
CANCEL sip:89113574005@89.28.166.183:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>
CSeq: 2 CANCEL
User-Agent: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 89.28.162.105:28952 (NAT)

<--- Reliably Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as3788c253
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 89.28.162.105:28952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf;received=89.28.162.105;rport=28952
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as3788c253
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
CSeq: 2 CANCEL
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 89.28.160.23:5060:
CANCEL sip:89113574005@89.28.160.23:5060 SIP/2.0
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
Max-Forwards: 70
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.7.1)
Content-Length: 0


---
Scheduling destruction of SIP dialog '57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:89.28.160.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>;tag=1048899395-3792784061-817677987-203481435
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 CANCEL
Contact: <sip:89113574005@89.28.160.23:5060>
Server: MERA MVTS3G v.4.4.0-16
Content-Length: 0

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

<--- SIP read from UDP:89.28.160.23:5060 --->
SIP/2.0 603 Request Terminated
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>;tag=1048899395-3792784061-817677987-203481435
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 INVITE
Contact: <sip:89113574005@89.28.160.23:5060>
Server: MERA MVTS3G v.4.4.0-16
Reason: SIP;cause=487;text="Request Terminated"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 89.28.160.23:5060:
ACK sip:89113574005@89.28.160.23:5060 SIP/2.0
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK44adc39b
Max-Forwards: 70
From: "69801" <sip:69801@89.28.166.183>;tag=as49b203f3
To: <sip:89113574005@89.28.160.23:5060>;tag=1048899395-3792784061-817677987-203481435
Contact: <sip:69801@89.28.166.183:5060>
Call-ID: 57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.7.1)
Content-Length: 0


---
Really destroying SIP dialog '57cece2e51a2f20d36acb6eb39c25092@89.28.166.183:5060' Method: INVITE

<--- SIP read from UDP:89.28.162.105:28952 --->
ACK sip:89113574005@89.28.166.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.222.42:5060;branch=z9hG4bKs82dd2a1bf
Max-Forwards: 70
Call-ID: 2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42
From: "Nigga" <sip:69777@89.28.166.183>;tag=2003756211
To: <sip:89113574005@89.28.166.183>;tag=as3788c253
Authorization: Digest realm="asterisk",nonce="138a695a",algorithm=MD5,uri="sip:89113574005@89.28.166.183:5060",username="69777",response="0777df44c6d0b1136bf718f58168ec7c"
CSeq: 2 ACK
User-Agent: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2e692666-2f6a1e19866eaf024b850080f0d4472e@172.22.222.42' Method: ACK
Reliably Transmitting (NAT) to 89.28.162.105:28952:
OPTIONS sip:69777@172.22.222.42:5060 SIP/2.0
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK027f24f3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@89.28.166.183>;tag=as4727204a
To: <sip:69777@172.22.222.42:5060>
Contact: <sip:Unknown@89.28.166.183:5060>
Call-ID: 368819e325de22bd34ca03f47b5a71f7@89.28.166.183:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Mon, 13 May 2013 14:08:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:89.28.162.105:28952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.166.183:5060;branch=z9hG4bK027f24f3;rport=5060;received=89.28.166.183
Call-ID: 368819e325de22bd34ca03f47b5a71f7@89.28.166.183:5060
From: "Unknown" <sip:Unknown@89.28.166.183>;tag=as4727204a
To: <sip:69777@172.22.222.42:5060>;tag=3613825629
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
Contact: <sip:172.22.222.42:5060>
Server: Panasonic_KX-UT123RU/01.167 (0080f0d4472e)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '368819e325de22bd34ca03f47b5a71f7@89.28.166.183:5060' Method: OPTIONS
89-28-166-183*CLI>

asterisk.ru

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