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Звонок на мобильный, как понять что номер не доступен?

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
rimaSamir
Сообщения: 4
Зарегистрирован: 31 окт 2017, 11:45

Звонок на мобильный, как понять что номер не доступен?

Сообщение rimaSamir »

Добрый день,

Звоню на мобильные номера с софтфона. Если номер не доступен или находится вне сети или не существует - оператором GSM проигрывается соответствующее сообщение. На астериске подвисает Session Progress. В конце Service Unavailable.

Еще одна проблема, когда мобильный оператор доет отбой не подняв трубку - на астериске так же подвисает Session Progress.

Подскажите пожалуйста куда копать?!
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Звонок на мобильный, как понять что номер не доступен?

Сообщение awsswa »

не пользоваться Session Progress ?

PS
посмотреть прилетает ли от оператор BYE по завершению проигрывания сообщения.
Понять - почему на BYE ваша АТС не хочет делать завершение звонка
Всякое общение без RTP трафика автоматически закрывается через 30 секунд
платный суппорт по мере возможностей
Вернуться к началу
ded
Сообщения: 15629
Зарегистрирован: 26 авг 2010, 19:00

Re: Звонок на мобильный, как понять что номер не доступен?

Сообщение ded »

Предлагаем к установке на Астериск 11-13 сложное решение channel_LEG, по определению голосовых сообщений GSM-провайдеров в предответном состоянии - "Абонент находится вне зоны или отключён..", "Абонент ведёт разговор по другой линии", с набором сэмплов для Теле2, Билайн, МТС и Мегафон, с возможностью добавления любых других голосовых сэмплов для распознавания.

channel_LEG слушает плечо оператора связи и, найдя соответствие, проставляет коды отбоя (Release complete = 18, =3), какие вам нужно.
Вернуться к началу
rimaSamir
Сообщения: 4
Зарегистрирован: 31 окт 2017, 11:45

Re: Звонок на мобильный, как понять что номер не доступен?

Сообщение rimaSamir »

Сколько будет стоить это чудо?
Вернуться к началу
rimaSamir
Сообщения: 4
Зарегистрирован: 31 окт 2017, 11:45

Re: Звонок на мобильный, как понять что номер не доступен?

Сообщение rimaSamir »

awsswa писал(а):не пользоваться Session Progress ?

PS
посмотреть прилетает ли от оператор BYE по завершению проигрывания сообщения.
Понять - почему на BYE ваша АТС не хочет делать завершение звонка
Всякое общение без RTP трафика автоматически закрывается через 30 секунд
Кстати версия FreePBX 14.0.1.1

добавил такие строчки, результат не изменился
prematuremedia = no
progressinband = yes

насчет 30 секунд - так оно и есть.
в основном мне надо разобраться с тем, почему когда не подняв трудку с мобильника дают отбой - ничего не приходит

SIP log


freepbx*CLI>
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from TCP:10.253.106.2:10312 --->
INVITE sip:10502918882@192.168.90.180 SIP/2.0
Via: SIP/2.0/TCP 0.0.0.0:10312;rport;branch=z9hG4bKPj7ac2328cac324c47b85629555f5dec05;alias
Max-Forwards: 70
From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba
To: <sip:10502918882@192.168.90.180>
Contact: "4441" <sip:4441@10.253.106.2:51683;ob>
Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30
CSeq: 23858 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.15.10
Content-Type: application/sdp
Content-Length: 903

v=0
o=- 3719224551 3719224551 IN IP4 10.253.106.2
s=pjmedia
b=AS:1498
t=0 0
a=X-nat:0
m=audio 4026 RTP/AVP 123 8 0 9 18 3 120 97 119 117 110 108 107 100 96 11 10 118 126 124 125 101
c=IN IP4 10.253.106.2
b=TIAS:1411200
a=rtcp:4027 IN IP4 10.253.106.2
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:120 AMR/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:119 speex/32000
a=rtpmap:117 speex/16000
a=rtpmap:110 speex/8000
a=rtpmap:108 SILK/24000
a=rtpmap:107 SILK/16000
a=rtpmap:100 SILK/12000
a=rtpmap:96 SILK/8000
a=rtpmap:11 L16/44100
a=rtpmap:10 L16/44100/2
a=rtpmap:118 L16/16000
a=rtpmap:126 L16/16000/2
a=rtpmap:124 L16/8000
a=rtpmap:125 L16/8000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 37 lines) ---
Sending to 10.253.106.2:10312 (no NAT)
Sending to 10.253.106.2:10312 (no NAT)

freepbx*CLI>
Using INVITE request as basis request - cd8299cb4d1b422a9bc2a850a35a7e30
Found peer '4441' for '4441' from 10.253.106.2:10312

<--- Reliably Transmitting (no NAT) to 10.253.106.2:10312 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/TCP 0.0.0.0:10312;branch=z9hG4bKPj7ac2328cac324c47b85629555f5dec05;alias;received=10.253.106.2;rport=10312

From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba

To: <sip:10502918882@192.168.90.180>;tag=as587f38a4

Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30

CSeq: 23858 INVITE

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6011d1e1"

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog 'cd8299cb4d1b422a9bc2a850a35a7e30' in 6400 ms (Method: INVITE)

freepbx*CLI>

<--- SIP read from TCP:10.253.106.2:10312 --->
ACK sip:10502918882@192.168.90.180 SIP/2.0
Via: SIP/2.0/TCP 0.0.0.0:10312;rport;branch=z9hG4bKPj7ac2328cac324c47b85629555f5dec05;alias
Max-Forwards: 70
From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba
To: <sip:10502918882@192.168.90.180>;tag=as587f38a4
Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30
CSeq: 23858 ACK
Content-Length: 0

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

freepbx*CLI>

<--- SIP read from TCP:10.253.106.2:10312 --->
INVITE sip:10502918882@192.168.90.180 SIP/2.0
Via: SIP/2.0/TCP 10.253.106.2:10312;rport;branch=z9hG4bKPj169da0c9cf8846d7b957629307c44179;alias
Max-Forwards: 70
From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba
To: <sip:10502918882@192.168.90.180>
Contact: "4441" <sip:4441@10.253.106.2:51683;ob>
Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30
CSeq: 23859 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.15.10
Authorization: Digest username="4441", realm="asterisk", nonce="6011d1e1", uri="sip:10502918882@192.168.90.180", response="8cd503d6bff25268e10aa8da2ef924a4", algorithm=MD5
Content-Type: application/sdp
Content-Length: 903

v=0
o=- 3719224551 3719224551 IN IP4 10.253.106.2
s=pjmedia
b=AS:1498
t=0 0
a=X-nat:0
m=audio 4026 RTP/AVP 123 8 0 9 18 3 120 97 119 117 110 108 107 100 96 11 10 118 126 124 125 101
c=IN IP4 10.253.106.2
b=TIAS:1411200
a=rtcp:4027 IN IP4 10.253.106.2
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:120 AMR/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:119 speex/32000
a=rtpmap:117 speex/16000
a=rtpmap:110 speex/8000
a=rtpmap:108 SILK/24000
a=rtpmap:107 SILK/16000
a=rtpmap:100 SILK/12000
a=rtpmap:96 SILK/8000
a=rtpmap:11 L16/44100
a=rtpmap:10 L16/44100/2
a=rtpmap:118 L16/16000
a=rtpmap:126 L16/16000/2
a=rtpmap:124 L16/8000
a=rtpmap:125 L16/8000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 37 lines) ---
Sending to 10.253.106.2:10312 (no NAT)
Using INVITE request as basis request - cd8299cb4d1b422a9bc2a850a35a7e30
Found peer '4441' for '4441' from 10.253.106.2:10312

freepbx*CLI>
Found RTP audio format 123
Found RTP audio format 8
Found RTP audio format 0

freepbx*CLI>
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 120
Found RTP audio format 97
Found RTP audio format 119
Found RTP audio format 117
Found RTP audio format 110
Found RTP audio format 108
Found RTP audio format 107
Found RTP audio format 100
Found RTP audio format 96
Found RTP audio format 11
Found RTP audio format 10
Found RTP audio format 118
Found RTP audio format 126
Found RTP audio format 124
Found RTP audio format 125
Found RTP audio format 101
Found audio description format opus for ID 123
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found unknown media description format AMR for ID 120
Found audio description format iLBC for ID 97
Found audio description format speex for ID 119
Found audio description format speex for ID 117
Found audio description format speex for ID 110
Found audio description format SILK for ID 108
Found audio description format SILK for ID 107
Found audio description format SILK for ID 100
Found audio description format SILK for ID 96
Found unknown media description format L16 for ID 11
Found unknown media description format L16 for ID 10
Found audio description format L16 for ID 118
Found audio description format L16 for ID 126
Found audio description format L16 for ID 124
Found audio description format L16 for ID 125
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|g729|silk8|ilbc|silk12|silk16|silk24|speex|speex16|slin16|speex32|slin24|slin|slin44|slin48)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.253.106.2:4026
Looking for 10502918882 in from-internal (domain 192.168.90.180)

freepbx*CLI>
sip_route_dump: route/path hop: <sip:4441@10.253.106.2:51683;ob>

freepbx*CLI>

<--- Transmitting (no NAT) to 10.253.106.2:10312 --->
SIP/2.0 100 Trying

Via: SIP/2.0/TCP 10.253.106.2:10312;branch=z9hG4bKPj169da0c9cf8846d7b957629307c44179;alias;received=10.253.106.2;rport=10312

From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba

To: <sip:10502918882@192.168.90.180>

Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30

CSeq: 23859 INVITE

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:10502918882@192.168.90.180:5060;transport=tcp>

Content-Length: 0




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

freepbx*CLI>
Audio is at 14022
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.109:5060:
INVITE sip:10502918882@192.168.50.109 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK0e88c90d

Max-Forwards: 70

From: <sip:3770707@192.168.90.180>;tag=as5e855674

To: <sip:10502918882@192.168.50.109>

Contact: <sip:3770707@192.168.90.180:5060>

Call-ID: 675aa94c3647b86666fdc0be1c8b978d@192.168.90.180:5060

CSeq: 102 INVITE

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:55:52 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 277



v=0

o=root 302167331 302167331 IN IP4 192.168.90.180

s=Asterisk PBX 14.6.0

c=IN IP4 192.168.90.180

t=0 0

m=audio 14022 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK0e88c90d
From: <sip:3770707@192.168.90.180>;tag=as5e855674
To: <sip:10502918882@192.168.50.109>;tag=1930388631
Call-ID: 675aa94c3647b86666fdc0be1c8b978d@192.168.90.180:5060
CSeq: 102 INVITE
Contact: <sip:1616@192.168.50.109:5060>
User-Agent: dble
Content-Length: 0

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

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK0e88c90d
From: <sip:3770707@192.168.90.180>;tag=as5e855674
To: <sip:10502918882@192.168.50.109>;tag=1930388631
Call-ID: 675aa94c3647b86666fdc0be1c8b978d@192.168.90.180:5060
CSeq: 102 INVITE
Contact: <sip:1616@192.168.50.109:5060>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1616@192.168.50.109:5060>

freepbx*CLI>

<--- Transmitting (no NAT) to 10.253.106.2:10312 --->
SIP/2.0 180 Ringing

Via: SIP/2.0/TCP 10.253.106.2:10312;branch=z9hG4bKPj169da0c9cf8846d7b957629307c44179;alias;received=10.253.106.2;rport=10312

From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba

To: <sip:10502918882@192.168.90.180>;tag=as01aed1a9

Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30

CSeq: 23859 INVITE

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:10502918882@192.168.90.180:5060;transport=tcp>

Content-Length: 0




<------------>
Audio is at 17966
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 10.253.106.2:10312 --->
SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 10.253.106.2:10312;branch=z9hG4bKPj169da0c9cf8846d7b957629307c44179;alias;received=10.253.106.2;rport=10312

From: "4441" <sip:4441@192.168.90.180>;tag=67871935d58049318f9f439bef5661ba

To: <sip:10502918882@192.168.90.180>;tag=as01aed1a9

Call-ID: cd8299cb4d1b422a9bc2a850a35a7e30

CSeq: 23859 INVITE

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:10502918882@192.168.90.180:5060;transport=tcp>

Content-Type: application/sdp

Require: timer

Content-Length: 324



v=0

o=root 643300683 643300683 IN IP4 192.168.90.180

s=Asterisk PBX 14.6.0

c=IN IP4 192.168.90.180

t=0 0

m=audio 17966 RTP/AVP 0 8 3 9 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:9 G722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


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

freepbx*CLI>

<--- SIP read from UDP:192.168.50.110:61016 --->

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

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK0e88c90d
From: <sip:3770707@192.168.90.180>;tag=as5e855674
To: <sip:10502918882@192.168.50.109>;tag=1930388631
Call-ID: 675aa94c3647b86666fdc0be1c8b978d@192.168.90.180:5060
CSeq: 102 INVITE
Contact: <sip:1616@192.168.50.109:5060>
User-Agent: dble
Content-Type: application/sdp
Content-Length: 236

v=0
o=dble 1510221352 1510221352 IN IP4 192.168.50.109
s=dble
c=IN IP4 192.168.50.109
t=0 0
m=audio 16384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (10 headers 11 lines) ---
sip_route_dump: route/path hop: <sip:1616@192.168.50.109:5060>

freepbx*CLI>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.50.109:16384

freepbx*CLI>

<--- SIP read from UDP:192.168.50.102:52180 --->

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

freepbx*CLI>

<--- SIP read from UDP:10.253.106.2:51683 --->

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

freepbx*CLI>

<--- SIP read from UDP:192.168.13.2:5060 --->
OPTIONS sip:192.168.13.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.2:5060;branch=z9hG4bK3c01e2e7;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.13.2>;tag=as289ff8b7
To: <sip:192.168.13.11>
Contact: <sip:Unknown@192.168.13.2:5060>
Call-ID: 6bfa88c842be88957e5139fe4301275b@192.168.13.2:5060
CSeq: 102 OPTIONS
User-Agent: Cisco-SIPGateway/IOS-224.x
Date: Thu, 09 Nov 2017 09:55:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.13.2:5060 (no NAT)

freepbx*CLI>
Looking for s in from-sip-external (domain 192.168.13.11)

<--- Transmitting (no NAT) to 192.168.13.2:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.13.2:5060;branch=z9hG4bK3c01e2e7;received=192.168.13.2;rport=5060

From: "Unknown" <sip:Unknown@192.168.13.2>;tag=as289ff8b7

To: <sip:192.168.13.11>;tag=as028457fd

Call-ID: 6bfa88c842be88957e5139fe4301275b@192.168.13.2:5060

CSeq: 102 OPTIONS

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: <sip:192.168.13.11:5060>

Accept: application/sdp

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '6bfa88c842be88957e5139fe4301275b@192.168.13.2:5060' in 32000 ms (Method: OPTIONS)

freepbx*CLI>

<--- SIP read from UDP:192.168.50.112:58266 --->

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

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.202:5060:
OPTIONS sip:5552@192.168.50.202:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK7896f050

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as1cd5a84a

To: <sip:5552@192.168.50.202:5060>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 10a7db431dad5cfa10e122690afd835a@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:03 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK7896f050
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as1cd5a84a
To: <sip:5552@192.168.50.202:5060>;tag=627475768
Call-ID: 10a7db431dad5cfa10e122690afd835a@192.168.90.180:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '10a7db431dad5cfa10e122690afd835a@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
REGISTER sip:192.168.90.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.109:5060;branch=z9hG4bK1157920722
From: "1616" <sip:1616@192.168.90.180>;tag=1606935673
To: "1616" <sip:1616@192.168.90.180>
Call-ID: 579122601@192.168.50.109
CSeq: 446 REGISTER
Contact: <sip:1616@192.168.50.109:5060>;expires=60
Authorization: Digest username="1616", realm="asterisk", nonce="5210514e", uri="sip:192.168.90.180", response="a2d241e6a536b9425aeeb7e81e157345", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.50.109:5060 (no NAT)
[2017-11-09 09:56:04] NOTICE[24688]: chan_sip.c:17335 check_auth: Correct auth, but based on stale nonce received from '"1616" <sip:1616@192.168.90.180>;tag=1606935673'

<--- Transmitting (no NAT) to 192.168.50.109:5060 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.50.109:5060;branch=z9hG4bK1157920722;received=192.168.50.109

From: "1616" <sip:1616@192.168.90.180>;tag=1606935673

To: "1616" <sip:1616@192.168.90.180>;tag=as05d0930c

Call-ID: 579122601@192.168.50.109

CSeq: 446 REGISTER

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b2065bb", stale=true

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '579122601@192.168.50.109' in 32000 ms (Method: REGISTER)

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
REGISTER sip:192.168.90.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.109:5060;branch=z9hG4bK761793427
From: "1616" <sip:1616@192.168.90.180>;tag=1606935673
To: "1616" <sip:1616@192.168.90.180>
Call-ID: 579122601@192.168.50.109
CSeq: 447 REGISTER
Contact: <sip:1616@192.168.50.109:5060>;expires=60
Authorization: Digest username="1616", realm="asterisk", nonce="2b2065bb", uri="sip:192.168.90.180", response="ff58de7e54c101b6695823ca7853d7a2", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.50.109:5060 (no NAT)

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.109:5060:
OPTIONS sip:1616@192.168.50.109:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK4ea9e995

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as5edebc1e

To: <sip:1616@192.168.50.109:5060>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 41e8e8e52f054323134dadd57752c197@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:04 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

<--- Transmitting (no NAT) to 192.168.50.109:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.50.109:5060;branch=z9hG4bK761793427;received=192.168.50.109

From: "1616" <sip:1616@192.168.90.180>;tag=1606935673

To: "1616" <sip:1616@192.168.90.180>;tag=as05d0930c

Call-ID: 579122601@192.168.50.109

CSeq: 447 REGISTER

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Expires: 60

Contact: <sip:1616@192.168.50.109:5060>;expires=60

Date: Thu, 09 Nov 2017 09:56:04 GMT

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '579122601@192.168.50.109' in 32000 ms (Method: REGISTER)

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK4ea9e995
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as5edebc1e
To: <sip:1616@192.168.50.109:5060>;tag=901162762
Call-ID: 41e8e8e52f054323134dadd57752c197@192.168.90.180:5060
CSeq: 102 OPTIONS
User-Agent: dble
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE
Content-Length: 0

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

freepbx*CLI>
Really destroying SIP dialog '41e8e8e52f054323134dadd57752c197@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>

<--- SIP read from UDP:192.168.50.102:52180 --->
REGISTER sip:192.168.90.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.102:52180;rport;branch=z9hG4bKPj452d9a47360a43b08246c6e05588e3b1
Max-Forwards: 70
From: "Fatima" <sip:5567@192.168.90.180>;tag=ff6c928d9d0f4b19a9fa4d05e7a4e9c6
To: "Fatima" <sip:5567@192.168.90.180>
Call-ID: f1595a757c774dbca7f47faf08076229
CSeq: 7417 REGISTER
User-Agent: MicroSIP/3.16.1
Contact: "Fatima" <sip:5567@192.168.50.102:52180;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

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

freepbx*CLI>
Sending to 192.168.50.102:52180 (no NAT)
Sending to 192.168.50.102:52180 (no NAT)

<--- Transmitting (no NAT) to 192.168.50.102:52180 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.50.102:52180;branch=z9hG4bKPj452d9a47360a43b08246c6e05588e3b1;received=192.168.50.102;rport=52180

From: "Fatima" <sip:5567@192.168.90.180>;tag=ff6c928d9d0f4b19a9fa4d05e7a4e9c6

To: "Fatima" <sip:5567@192.168.90.180>;tag=as6f124b82

Call-ID: f1595a757c774dbca7f47faf08076229

CSeq: 7417 REGISTER

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cf8dd3b"

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog 'f1595a757c774dbca7f47faf08076229' in 32000 ms (Method: REGISTER)

freepbx*CLI>

<--- SIP read from UDP:192.168.50.102:52180 --->
REGISTER sip:192.168.90.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.102:52180;rport;branch=z9hG4bKPj089d73f529e54d0bb5b0f6b4273e6470
Max-Forwards: 70
From: "Fatima" <sip:5567@192.168.90.180>;tag=ff6c928d9d0f4b19a9fa4d05e7a4e9c6
To: "Fatima" <sip:5567@192.168.90.180>
Call-ID: f1595a757c774dbca7f47faf08076229
CSeq: 7418 REGISTER
User-Agent: MicroSIP/3.16.1
Contact: "Fatima" <sip:5567@192.168.50.102:52180;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="5567", realm="asterisk", nonce="6cf8dd3b", uri="sip:192.168.90.180", response="cc252db58da12a23614dd8943a8ac9bd", algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.50.102:52180 (no NAT)

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.102:52180:
OPTIONS sip:5567@192.168.50.102:52180;ob SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK1e0b5cfd

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as623d5efd

To: <sip:5567@192.168.50.102:52180;ob>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 356867ec43a9c8b067215a5a291dfcf7@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:08 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

<--- Transmitting (no NAT) to 192.168.50.102:52180 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.50.102:52180;branch=z9hG4bKPj089d73f529e54d0bb5b0f6b4273e6470;received=192.168.50.102;rport=52180

From: "Fatima" <sip:5567@192.168.90.180>;tag=ff6c928d9d0f4b19a9fa4d05e7a4e9c6

To: "Fatima" <sip:5567@192.168.90.180>;tag=as6f124b82

Call-ID: f1595a757c774dbca7f47faf08076229

CSeq: 7418 REGISTER

Server: FPBX-14.0.1.1(14.6.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Expires: 300

Contact: <sip:5567@192.168.50.102:52180;ob>;expires=300

Date: Thu, 09 Nov 2017 09:56:08 GMT

Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '77bf59d01d177c4d5098c7aa20d2d3de@192.168.90.180:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.50.102:52180:
NOTIFY sip:5567@192.168.50.102:52180;ob SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK293e85d0

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as46b7721c

To: <sip:5567@192.168.50.102:52180;ob>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 77bf59d01d177c4d5098c7aa20d2d3de@192.168.90.180:5060

CSeq: 102 NOTIFY

User-Agent: FPBX-14.0.1.1(14.6.0)

Event: message-summary

Content-Type: application/simple-message-summary

Content-Length: 90



Messages-Waiting: yes

Message-Account: sip:*97@192.168.90.180

Voice-Message: 2/0 (0/0)


---
Scheduling destruction of SIP dialog 'f1595a757c774dbca7f47faf08076229' in 32000 ms (Method: REGISTER)

freepbx*CLI>

<--- SIP read from UDP:192.168.50.102:52180 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;received=192.168.90.180;branch=z9hG4bK1e0b5cfd
Call-ID: 356867ec43a9c8b067215a5a291dfcf7@192.168.90.180:5060
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as623d5efd
To: <sip:5567@192.168.50.102;ob>;tag=z9hG4bK1e0b5cfd
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.16.1
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '356867ec43a9c8b067215a5a291dfcf7@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>

<--- SIP read from UDP:192.168.50.102:52180 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;received=192.168.90.180;branch=z9hG4bK293e85d0
Call-ID: 77bf59d01d177c4d5098c7aa20d2d3de@192.168.90.180:5060
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as46b7721c
To: <sip:5567@192.168.50.102;ob>;tag=z9hG4bK293e85d0
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '77bf59d01d177c4d5098c7aa20d2d3de@192.168.90.180:5060' Method: NOTIFY

freepbx*CLI>

<--- SIP read from UDP:192.168.50.110:61016 --->

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

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.110:61016:
OPTIONS sip:5566@192.168.50.110:61016;ob SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK41509a1f

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as428aa3d3

To: <sip:5566@192.168.50.110:61016;ob>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 6918f8301b26152b47fdd15a2efece1f@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:11 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.110:61016 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;received=192.168.90.180;branch=z9hG4bK41509a1f
Call-ID: 6918f8301b26152b47fdd15a2efece1f@192.168.90.180:5060
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as428aa3d3
To: <sip:5566@192.168.50.110;ob>;tag=z9hG4bK41509a1f
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.16.1
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '6918f8301b26152b47fdd15a2efece1f@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.201:5060:
OPTIONS sip:5556@192.168.50.201:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK7de521fa

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as082ad72a

To: <sip:5556@192.168.50.201:5060>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 7b09d7a61aebb1cb54a1430427c1f3db@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:11 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK7de521fa
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as082ad72a
To: <sip:5556@192.168.50.201:5060>;tag=2138426417
Call-ID: 7b09d7a61aebb1cb54a1430427c1f3db@192.168.90.180:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7b09d7a61aebb1cb54a1430427c1f3db@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.203:5060:
OPTIONS sip:5553@192.168.50.203:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK20c7987d

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as20732dd2

To: <sip:5553@192.168.50.203:5060>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 11c24196238cb198634aa2c84f02b219@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:12 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.203:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK20c7987d
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as20732dd2
To: <sip:5553@192.168.50.203:5060>;tag=1649586865
Call-ID: 11c24196238cb198634aa2c84f02b219@192.168.90.180:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '11c24196238cb198634aa2c84f02b219@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>

<--- SIP read from UDP:10.253.106.2:51683 --->

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

freepbx*CLI>
Reliably Transmitting (no NAT) to 10.253.106.2:51683:
OPTIONS sip:4441@10.253.106.2:51683;ob SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK016d26bf

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as5a379c7e

To: <sip:4441@10.253.106.2:51683;ob>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 4e1bf1c05d5f3b72447d3dbc07b3ae4d@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:13 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:10.253.106.2:51683 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;received=192.168.90.180;branch=z9hG4bK016d26bf
Call-ID: 4e1bf1c05d5f3b72447d3dbc07b3ae4d@192.168.90.180:5060
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as5a379c7e
To: <sip:4441@10.253.106.2;ob>;tag=z9hG4bK016d26bf
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.15.10
Content-Length: 0

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

freepbx*CLI>
--- (12 headers 0 lines) ---
Really destroying SIP dialog '4e1bf1c05d5f3b72447d3dbc07b3ae4d@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>

<--- SIP read from UDP:192.168.50.112:58266 --->

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

freepbx*CLI>

<--- SIP read from UDP:192.168.50.109:5060 --->

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

freepbx*CLI>
Reliably Transmitting (no NAT) to 192.168.50.200:5060:
OPTIONS sip:5554@192.168.50.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK211487cb

Max-Forwards: 70

From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as2cb586e3

To: <sip:5554@192.168.50.200:5060>

Contact: <sip:Unknown@192.168.90.180:5060>

Call-ID: 470c56561fde436b2674bcb41247e1ee@192.168.90.180:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.1.1(14.6.0)

Date: Thu, 09 Nov 2017 09:56:18 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0




---

freepbx*CLI>

<--- SIP read from UDP:192.168.50.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.90.180:5060;branch=z9hG4bK211487cb
From: "Unknown" <sip:Unknown@192.168.90.180>;tag=as2cb586e3
To: <sip:5554@192.168.50.200:5060>;tag=1045791962
Call-ID: 470c56561fde436b2674bcb41247e1ee@192.168.90.180:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '470c56561fde436b2674bcb41247e1ee@192.168.90.180:5060' Method: OPTIONS

freepbx*CLI>
[/quote]

вот есть такой лог, это уже когда отбой дает абонент
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
[2017-11-09 09:44:32] DEBUG[32160][C-00000006]: res_rtp_asterisk.c:4558 ast_rtcp_interpret: Got RTCP report of 84 bytes

freepbx*CLI>
[2017-11-09 09:44:32] DEBUG[32262]: manager.c:6363 process_message: Running action 'Login'

freepbx*CLI>
[2017-11-09 09:44:33] DEBUG[24688]: chan_sip.c:16940 parse_register_contact: Store REGISTER's Contact header for call routing.

freepbx*CLI>
[2017-11-09 09:44:33] DEBUG[24688]: chan_sip.c:9017 __sip_alloc: Allocating new SIP dialog for 455b7c10121fda0e54b4f2e73a9b4f96@127.0.0.1:5060 - OPTIONS (No RTP)

freepbx*CLI>
[2017-11-09 09:44:33] DEBUG[24688]: chan_sip.c:8806 change_callid_pvt: SIP call-id changed from '455b7c10121fda0e54b4f2e73a9b4f96@127.0.0.1:5060' to '468d989e247a5b454f1c1d9b0a28d359@192.168.90.180:5060'
[2017-11-09 09:44:33] DEBUG[24688]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method OPTIONS - callid 468d989e247a5b454f1c1d9b0a28d359@192.168.90.180:5060

freepbx*CLI>
[2017-11-09 09:44:33] DEBUG[24688]: chan_sip.c:4543 __sip_ack: Stopping retransmission on '468d989e247a5b454f1c1d9b0a28d359@192.168.90.180:5060' of Request 102: Match Found

freepbx*CLI>
[2017-11-09 09:44:37] DEBUG[32160][C-00000006]: res_rtp_asterisk.c:4558 ast_rtcp_interpret: Got RTCP report of 84 bytes

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:9017 __sip_alloc: Allocating new SIP dialog for 87e4cd38d0ec49b59aca9cb363fd71f1 - REGISTER (No RTP)

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:16962 parse_register_contact: Store REGISTER's src-IP:port for call routing.

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:9017 __sip_alloc: Allocating new SIP dialog for 2022749b18bf47a64032867351c5a037@127.0.0.1:5060 - OPTIONS (No RTP)

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:8806 change_callid_pvt: SIP call-id changed from '2022749b18bf47a64032867351c5a037@127.0.0.1:5060' to '4a6c9f396f9622a532d355c76cba73da@192.168.90.180:5060'
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method OPTIONS - callid 4a6c9f396f9622a532d355c76cba73da@192.168.90.180:5060

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:9017 __sip_alloc: Allocating new SIP dialog for 0bdda18173580ccf2472ce60112fa44a@127.0.0.1:5060 - NOTIFY (No RTP)

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:8806 change_callid_pvt: SIP call-id changed from '0bdda18173580ccf2472ce60112fa44a@127.0.0.1:5060' to '00d18fb8608bf0995a745e54259cf481@192.168.90.180:5060'
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:3402 initialize_initreq: Initializing initreq for method NOTIFY - callid 00d18fb8608bf0995a745e54259cf481@192.168.90.180:5060

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:4543 __sip_ack: Stopping retransmission on '4a6c9f396f9622a532d355c76cba73da@192.168.90.180:5060' of Request 102: Match Found

freepbx*CLI>
[2017-11-09 09:44:39] DEBUG[24688]: chan_sip.c:4543 __sip_ack: Stopping retransmission on '00d18fb8608bf0995a745e54259cf481@192.168.90.180:5060' of Request 102: Match Found

freepbx*CLI>
[2017-11-09 09:44:42] DEBUG[32160][C-00000006]: res_rtp_asterisk.c:4558 ast_rtcp_interpret: Got RTCP report of 84 bytes

freepbx*CLI>
Портянки. причте под споллер !!!!
Вернуться к началу
ded
Сообщения: 15629
Зарегистрирован: 26 авг 2010, 19:00

Re: Звонок на мобильный, как понять что номер не доступен?

Сообщение ded »

rimaSamir писал(а):Сколько будет стоить это чудо?
Самир, Вам по эл. почте направил коллега предложение, но Вы молчите.
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