VIDEOCHAT  ::   FAQ  ::   Поиск  ::   Регистрация  ::   Вход

Нет голоса у внешних клиентов

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

Модераторы: april22, Zavr2008

avsnt
Сообщения: 6
Зарегистрирован: 20 дек 2016, 09:27

Нет голоса у внешних клиентов

Сообщение avsnt »

Доброго дня! Имеем FreePBX Distro с 13 астером на борту. Практически во всем получилось разобраться, вот только одна проблема осталась.
Для начала немного предыстории. Все было успешно настроено и работало за исключением одного провайдера. Пройдет регистрация, через несколько минут транк отпадает, шлет 401, потом опять регистрация проходит, и вновь по кругу. Раньше стояла 3CX и проблем с этим провайдером не было. Сравнили пакеты с 3CX и астериска - разница только в VIA. 3CX шлет внутренний ip, а астер - внешний. Окей, поменял externip на внутренний адрес, проблема с провайдером решена, а вот у внешних sip-телефонов пропал голос. Астер находится за nat, проброс портов 5060 и 10000-20000 сделан. VPN возможности поднять нет.
Что же делать? Как же быть, чтобы и транк от провайдера и внешние клиенты работали?
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: sipconf
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
useragent=FPBX-13.0.190.7(13.12.1)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
rtpend=20000
rtpstart=10000
tcpenable=no
callevents=yes
bindport=5060
jbenable=no
notifyringing=yes
allowguest=no
tlsbindaddr=[::]:5061
tlsclientmethod=sslv2
g726nonstandard=no
srvlookup=no
tlsenable=no
defaultexpiry=360
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=360
notifyhold=yes
checkmwi=10
nat=force_rport,comedia
externip=192.168.20.14
ALLOW_SIP_ANON=no
callerid=Unknown
localnet=192.168.20.0/24
localnet=192.168.40.0/24
localnet=172.16.250.0/24
language=ru
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: extension
[210]
deny=0.0.0.0/0.0.0.0
secret=mypass
dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/210
permit=0.0.0.0/0.0.0.0
callerid=ДДВ Дом <210>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
192.168.20.14 - астериск
*.*.11.246 - внешний адрес клиента
*.*.97.133 - внешний адрес астериска
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: debug
<--- SIP read from UDP:*.*.11.246:5060 --->
REGISTER sip:192.168.20.14 SIP/2.0
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bKe3bc261c7074bdf955cc3ce21214999a;rport
From: "210" <sip:210@192.168.20.14>;tag=337712029
To: "210" <sip:210@192.168.20.14>
Call-ID: 1869058261@*_*_11_246
CSeq: 1 REGISTER
Contact: <sip:210@*.*.11.246:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to *.*.11.246:5060 (NAT)
Sending to *.*.11.246:5060 (NAT)

<--- Transmitting (NAT) to *.*.11.246:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bKe3bc261c7074bdf955cc3ce21214999a;received=*.*.11.246;rport=5060
From: "210" <sip:210@192.168.20.14>;tag=337712029
To: "210" <sip:210@192.168.20.14>;tag=as3b1f1b4a
Call-ID: 1869058261@*_*_11_246
CSeq: 1 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2092da25"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1869058261@*_*_11_246' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:*.*.11.246:5060 --->
REGISTER sip:192.168.20.14 SIP/2.0
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bKb2a88ba5c257bb4cb85f40c2492d80f;rport
From: "210" <sip:210@192.168.20.14>;tag=337712029
To: "210" <sip:210@192.168.20.14>
Call-ID: 1869058261@*_*_11_246
CSeq: 2 REGISTER
Contact: <sip:210@*.*.11.246:5060>
Authorization: Digest username="210", realm="asterisk", algorithm=MD5, uri="sip:192.168.20.14", nonce="2092da25", response="7736d68784853d5efac38cdaf46a76cd"
Max-Forwards: 70
User-Agent: A510 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to *.*.11.246:5060 (NAT)
Reliably Transmitting (NAT) to *.*.11.246:5060:
OPTIONS sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK7adb6cab;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as18b8f563
To: <sip:210@*.*.11.246:5060>
Contact: <sip:Unknown@192.168.20.14:5060>
Call-ID: 5a5745667ca29d5e4acee7505f782d61@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 06:48:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to *.*.11.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bKb2a88ba5c257bb4cb85f40c2492d80f;received=*.*.11.246;rport=5060
From: "210" <sip:210@192.168.20.14>;tag=337712029
To: "210" <sip:210@192.168.20.14>;tag=as3b1f1b4a
Call-ID: 1869058261@*_*_11_246
CSeq: 2 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 180
Contact: <sip:210@*.*.11.246:5060>;expires=180
Date: Tue, 20 Dec 2016 06:48:28 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1869058261@*_*_11_246' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK7adb6cab;rport=5060;received=*.*.97.133
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as18b8f563
To: <sip:210@*.*.11.246:5060>
Call-ID: 5a5745667ca29d5e4acee7505f782d61@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: A510 IP/42.075.00.000.000
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5a5745667ca29d5e4acee7505f782d61@192.168.20.14:5060' Method: OPTIONS

<--- SIP read from UDP:*.*.11.246:5060 --->


<------------->
Really destroying SIP dialog '1869058261@*_*_11_246' Method: REGISTER

<--- SIP read from UDP:*.*.11.246:5060 --->


<------------->
Reliably Transmitting (NAT) to *.*.11.246:5060:
OPTIONS sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK6fbf2555;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as0cd6548c
To: <sip:210@*.*.11.246:5060>
Contact: <sip:Unknown@192.168.20.14:5060>
Call-ID: 3bcfe24033d0b5884326074356be1b04@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 06:49:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:*.*.11.246:5060 --->


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

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK6fbf2555;rport=5060;received=*.*.97.133
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as0cd6548c
To: <sip:210@*.*.11.246:5060>;tag=ar1be7459b
Call-ID: 3bcfe24033d0b5884326074356be1b04@192.168.20.14:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: A510 IP/42.075.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '3bcfe24033d0b5884326074356be1b04@192.168.20.14:5060' Method: OPTIONS

<--- SIP read from UDP:*.*.11.246:5060 --->


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

<--- SIP read from UDP:*.*.11.246:5060 --->


<------------->
Audio is at 19530
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to *.*.11.246:5060:
INVITE sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK12e73295;rport
Max-Forwards: 70
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>
Contact: <sip:200@192.168.20.14:5060>
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 06:50:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Денис Катаев" <sip:200@192.168.20.14>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 625208598 625208598 IN IP4 192.168.20.14
s=Asterisk PBX 13.12.1
c=IN IP4 192.168.20.14
t=0 0
m=audio 19530 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK12e73295;rport=5060;received=*.*.97.133
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 102 INVITE
Contact: <sip:210@*.*.11.246:5060>
User-Agent: A510 IP/42.075.00.000.000
Content-Length: 0

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

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK12e73295;rport=5060;received=*.*.97.133
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 102 INVITE
Contact: <sip:210@*.*.11.246:5060>
Allow-Events: message-summary, refer, ua-profile
User-Agent: A510 IP/42.075.00.000.000
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:210@*.*.11.246:5060>
Reliably Transmitting (NAT) to *.*.11.246:5060:
OPTIONS sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK7e283af1;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as14688ff5
To: <sip:210@*.*.11.246:5060>
Contact: <sip:Unknown@192.168.20.14:5060>
Call-ID: 71f031ba4c4090565ad49bc31e85a89e@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 06:50:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK7e283af1;rport=5060;received=*.*.97.133
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as14688ff5
To: <sip:210@*.*.11.246:5060>
Call-ID: 71f031ba4c4090565ad49bc31e85a89e@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: A510 IP/42.075.00.000.000
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '71f031ba4c4090565ad49bc31e85a89e@192.168.20.14:5060' Method: OPTIONS

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK12e73295;rport=5060;received=*.*.97.133
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 102 INVITE
Contact: <sip:210@*.*.11.246:5060>
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 208

v=0
o=210 5004 1 IN IP4 *.*.11.246
s=Mapping
c=IN IP4 *.*.11.246
t=0 0
m=audio 5004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port *.*.11.246:5004
sip_route_dump: route/path hop: <sip:210@*.*.11.246:5060>
Transmitting (NAT) to *.*.11.246:5060:
ACK sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK239b2086;rport
Max-Forwards: 70
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Contact: <sip:200@192.168.20.14:5060>
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.190.7(13.12.1)
Content-Length: 0


---
Scheduling destruction of SIP dialog '1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to *.*.11.246:5060:
BYE sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK2f4bc207;rport
Max-Forwards: 70
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 103 BYE
User-Agent: FPBX-13.0.190.7(13.12.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:*.*.11.246:5060 --->
REGISTER sip:192.168.20.14 SIP/2.0
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bK8fa520162bed18759dc6c67897737b45;rport
From: "210" <sip:210@192.168.20.14>;tag=2826863242
To: "210" <sip:210@192.168.20.14>
Call-ID: 1869058261@*_*_11_246
CSeq: 3 REGISTER
Contact: <sip:210@*.*.11.246:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to *.*.11.246:5060 (NAT)
Sending to *.*.11.246:5060 (NAT)

<--- Transmitting (NAT) to *.*.11.246:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bK8fa520162bed18759dc6c67897737b45;received=*.*.11.246;rport=5060
From: "210" <sip:210@192.168.20.14>;tag=2826863242
To: "210" <sip:210@192.168.20.14>;tag=as797380be
Call-ID: 1869058261@*_*_11_246
CSeq: 3 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7fc3b72b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1869058261@*_*_11_246' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK2f4bc207;rport=5060;received=*.*.97.133
From: "Денис Катаев" <sip:200@192.168.20.14>;tag=as3be79d5b
To: <sip:210@*.*.11.246:5060>;tag=1362651614
Call-ID: 1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060
CSeq: 103 BYE
User-Agent: A510 IP/42.075.00.000.000
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1e84e93e6bfebf973ec1c5ec46db6582@192.168.20.14:5060' Method: INVITE

<--- SIP read from UDP:*.*.11.246:5060 --->
REGISTER sip:192.168.20.14 SIP/2.0
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bK7a4073ae9afca45b6e63d7e35400100f;rport
From: "210" <sip:210@192.168.20.14>;tag=2826863242
To: "210" <sip:210@192.168.20.14>
Call-ID: 1869058261@*_*_11_246
CSeq: 4 REGISTER
Contact: <sip:210@*.*.11.246:5060>
Authorization: Digest username="210", realm="asterisk", algorithm=MD5, uri="sip:192.168.20.14", nonce="7fc3b72b", response="bcb1493d77024e0494d68d15b960b775"
Max-Forwards: 70
User-Agent: A510 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to *.*.11.246:5060 (NAT)
Reliably Transmitting (NAT) to *.*.11.246:5060:
OPTIONS sip:210@*.*.11.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK4256b004;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as0fed60f1
To: <sip:210@*.*.11.246:5060>
Contact: <sip:Unknown@192.168.20.14:5060>
Call-ID: 057dce6d648b8f7e7e784a7d1c12f5dc@192.168.20.14:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 06:50:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to *.*.11.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.11.246:5060;branch=z9hG4bK7a4073ae9afca45b6e63d7e35400100f;received=*.*.11.246;rport=5060
From: "210" <sip:210@192.168.20.14>;tag=2826863242
To: "210" <sip:210@192.168.20.14>;tag=as797380be
Call-ID: 1869058261@*_*_11_246
CSeq: 4 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 180
Contact: <sip:210@*.*.11.246:5060>;expires=180
Date: Tue, 20 Dec 2016 06:50:43 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1869058261@*_*_11_246' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:*.*.11.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.14:5060;branch=z9hG4bK4256b004;rport=5060;received=*.*.97.133
From: "Unknown" <sip:Unknown@192.168.20.14>;tag=as0fed60f1
To: <sip:210@*.*.11.246:5060>;tag=ar1gde71g0
Call-ID: 057dce6d648b8f7e7e784a7d1c12f5dc@192.168.20.14:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: A510 IP/42.075.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '057dce6d648b8f7e7e784a7d1c12f5dc@192.168.20.14:5060' Method: OPTIONS
asterisker
Сообщения: 65
Зарегистрирован: 18 июл 2016, 11:40

Re: Нет голоса у внешних клиентов

Сообщение asterisker »

externip=192.168.20.14
*.*.97.133 - внешний адрес астериска
localnet=192.168.20.0/24
Synopsis
externip = extern.ip.address

Description
Indicates the IP address (alternatively you can enter a hostname) that will be used as the source IP address for all SIP messages when NAT is specified.
avsnt
Сообщения: 6
Зарегистрирован: 20 дек 2016, 09:27

Re: Нет голоса у внешних клиентов

Сообщение avsnt »

asterisker писал(а):externip=192.168.20.14
*.*.97.133 - внешний адрес астериска
localnet=192.168.20.0/24
Synopsis
externip = extern.ip.address

Description
Indicates the IP address (alternatively you can enter a hostname) that will be used as the source IP address for all SIP messages when NAT is specified.
Я это понимаю, но суть в том, что если там указать внешний адрес, то начнутся проблемы с транком, а внешние клиенту будут работать на ура.
Аватара пользователя
Zavr2008
Сообщения: 2250
Зарегистрирован: 27 янв 2011, 00:35
Контактная информация:

Re: Нет голоса у внешних клиентов

Сообщение Zavr2008 »

Пройдет регистрация, через несколько минут транк отпадает, шлет 401,
И что говорит оператор SIP на подобное?
Внутренний адрес подставлять - неправильно.

Точно ли 401 отдает, а не 403?

Выложите логи и sip отладку.
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1, настройка Asterisk/FreePBX, подключение CRM
avsnt
Сообщения: 6
Зарегистрирован: 20 дек 2016, 09:27

Re: Нет голоса у внешних клиентов

Сообщение avsnt »

Zavr2008 писал(а):
Пройдет регистрация, через несколько минут транк отпадает, шлет 401,
И что говорит оператор SIP на подобное?
Внутренний адрес подставлять - неправильно.

Точно ли 401 отдает, а не 403?

Выложите логи и sip отладку.
Провайдер грешит на нат.
Вернул внешний адрес.
debug

Код: Выделить всё

Reliably Transmitting (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK71faceac;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as73c65a39
To: <sip:230333@80.64.21.250>
Call-ID: 2733c4bd2fd69cab47a95120663245fc@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK71faceac;rport=58708;received=*.*.97.133
From: <sip:230333@80.64.21.250>;tag=as73c65a39
To: <sip:230333@80.64.21.250>;tag=2541c7e0c69c11e6a3c62c44fd933994
Call-ID: 2733c4bd2fd69cab47a95120663245fc@127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="42D3065E"
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name 80.64.21.250
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK5b2b0734;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as73c65a39
To: <sip:230333@80.64.21.250>
Call-ID: 2733c4bd2fd69cab47a95120663245fc@127.0.0.1
CSeq: 103 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Authorization: Digest username="230333", realm="SIP-REGISTRAR", algorithm=MD5, uri="sip:80.64.21.250", nonce="42D3065E", response="d5b9c0bdcf23ff90fb61b07f34bfbfc8"
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK5b2b0734;rport=58708;received=*.*.97.133
From: <sip:230333@80.64.21.250>;tag=as73c65a39
To: <sip:230333@80.64.21.250>;tag=254598dec69c11e6a3c62c44fd933994
Call-ID: 2733c4bd2fd69cab47a95120663245fc@127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:230333@*.*.97.133:5060>;expires=120
Expires: 120
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2733c4bd2fd69cab47a95120663245fc@127.0.0.1' Method: REGISTER

<--- SIP read from UDP:80.64.21.250:5060 --->
OPTIONS sip:*.*.97.133 SIP/2.0
Via: SIP/2.0/UDP 80.64.21.250:5060;rport;branch=z9hG4bK-2523130670-3859913926-1143785123-2486801405
From: <sip:80.64.21.250>;tag=2824923950-3859913926-1143785123-2486801405
To: <sip:*.*.97.133>
Call-ID: 2eeb60eec69c11e6a3c62c44fd933994@80.64.21.250
CSeq: 2 OPTIONS
Max-Forwards: 70
User-Agent: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 80.64.21.250:5060 (NAT)
Looking for s in from-sip-external (domain *.*.97.133)

<--- Transmitting (NAT) to 80.64.21.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.64.21.250:5060;branch=z9hG4bK-2523130670-3859913926-1143785123-2486801405;received=80.64.21.250;rport=5060
From: <sip:80.64.21.250>;tag=2824923950-3859913926-1143785123-2486801405
To: <sip:*.*.97.133>;tag=as5bfdfe6f
Call-ID: 2eeb60eec69c11e6a3c62c44fd933994@80.64.21.250
CSeq: 2 OPTIONS
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:*.*.97.133:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2eeb60eec69c11e6a3c62c44fd933994@80.64.21.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '2eeb60eec69c11e6a3c62c44fd933994@80.64.21.250' Method: OPTIONS
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #1 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Reliably Transmitting (NAT) to 80.64.21.250:5060:
OPTIONS sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK2168f830;rport
Max-Forwards: 70
From: "Unknown" <sip:230333@*.*.97.133>;tag=as4e0fac07
To: <sip:80.64.21.250>
Contact: <sip:230333@*.*.97.133:5060>
Call-ID: 64ab35ec79e46eb94b7e73374da98c50@*.*.97.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 10:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK2168f830;rport=55254;received=*.*.97.133
From: "Unknown" <sip:230333@*.*.97.133>;tag=as4e0fac07
To: <sip:80.64.21.250>;tag=1554871650-3859913926-1143785123-2486801405
Call-ID: 64ab35ec79e46eb94b7e73374da98c50@*.*.97.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
[2016-12-20 15:09:25] NOTICE[1802]: chan_sip.c:24444 handle_response_peerpoke: Peer 'Tagnet' is now Reachable. (10ms / 1000ms)
Really destroying SIP dialog '64ab35ec79e46eb94b7e73374da98c50@*.*.97.133:5060' Method: OPTIONS
Retransmitting #2 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #3 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #4 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #5 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #6 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #7 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #8 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #9 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
Retransmitting #10 (NAT) to 80.64.21.250:5060:
REGISTER sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK645a579f;rport
Max-Forwards: 70
From: <sip:230333@80.64.21.250>;tag=as1a0ca8e0
To: <sip:230333@80.64.21.250>
Call-ID: 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.190.7(13.12.1)
Expires: 360
Contact: <sip:230333@*.*.97.133:5060>
Content-Length: 0


---
[2016-12-20 15:09:56] WARNING[1802]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 1d6c39dc34653e0f114594f36ddaf5de@127.0.0.1 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

Reliably Transmitting (NAT) to 80.64.21.250:5060:
OPTIONS sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK1c27c574;rport
Max-Forwards: 70
From: "Unknown" <sip:230333@*.*.97.133>;tag=as1e43c21d
To: <sip:80.64.21.250>
Contact: <sip:230333@*.*.97.133:5060>
Call-ID: 059e2666706b72a5625de40328a0b8e9@*.*.97.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 10:10:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK1c27c574;rport=55254;received=*.*.97.133
From: "Unknown" <sip:230333@*.*.97.133>;tag=as1e43c21d
To: <sip:80.64.21.250>;tag=4101126790-3859913926-1143785123-2486801405
Call-ID: 059e2666706b72a5625de40328a0b8e9@*.*.97.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '059e2666706b72a5625de40328a0b8e9@*.*.97.133:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 80.64.21.250:5060:
OPTIONS sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK6be8c4e9;rport
Max-Forwards: 70
From: "Unknown" <sip:230333@*.*.97.133>;tag=as530badcd
To: <sip:80.64.21.250>
Contact: <sip:230333@*.*.97.133:5060>
Call-ID: 4f0bc4987d94b5d53517959c6cd8fbe3@*.*.97.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 10:11:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK6be8c4e9;rport=54783;received=*.*.97.133
From: "Unknown" <sip:230333@*.*.97.133>;tag=as530badcd
To: <sip:80.64.21.250>;tag=2384986025-3859913926-1143785123-2486801405
Call-ID: 4f0bc4987d94b5d53517959c6cd8fbe3@*.*.97.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '4f0bc4987d94b5d53517959c6cd8fbe3@*.*.97.133:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 80.64.21.250:5060:
OPTIONS sip:80.64.21.250 SIP/2.0
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK1cf6114d;rport
Max-Forwards: 70
From: "Unknown" <sip:230333@*.*.97.133>;tag=as61a3b50f
To: <sip:80.64.21.250>
Contact: <sip:230333@*.*.97.133:5060>
Call-ID: 17d049e573e956a506dd38376e3c39f7@*.*.97.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Tue, 20 Dec 2016 10:12:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK1cf6114d;rport=54783;received=*.*.97.133
From: "Unknown" <sip:230333@*.*.97.133>;tag=as61a3b50f
To: <sip:80.64.21.250>;tag=1622066125-3859913926-1143785123-2486801405
Call-ID: 17d049e573e956a506dd38376e3c39f7@*.*.97.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-20g
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '17d049e573e956a506dd38376e3c39f7@*.*.97.133:5060' Method: OPTIONS
Аватара пользователя
Zavr2008
Сообщения: 2250
Зарегистрирован: 27 янв 2011, 00:35
Контактная информация:

Re: Нет голоса у внешних клиентов

Сообщение Zavr2008 »

Может ему Ваш OPTIONS не нравится? qualify=no поставьте. И еще подождите минут 15-20 перед повторной регистрацией - походу банит он..

Еще странно:
realm="SIP-REGISTRAR"
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1, настройка Asterisk/FreePBX, подключение CRM
Аватара пользователя
Zavr2008
Сообщения: 2250
Зарегистрирован: 27 янв 2011, 00:35
Контактная информация:

Re: Нет голоса у внешних клиентов

Сообщение Zavr2008 »

Рецепт прост - нужно отключить транк в Астере, зарегить на софт-фоне X-lite и записать REGISTER и ответ на него в Wireshark.
Далее сравнивать.
Выложите дамп.
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1, настройка Asterisk/FreePBX, подключение CRM
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Нет голоса у внешних клиентов

Сообщение Vlad1983 »

avsnt писал(а):externip=192.168.20.14
...
*.*.97.133 - внешний адрес астериска
вся проблема в этом
ЛС: @rostel
Аватара пользователя
Zavr2008
Сообщения: 2250
Зарегистрирован: 27 янв 2011, 00:35
Контактная информация:

Re: Нет голоса у внешних клиентов

Сообщение Zavr2008 »

Он отключил это, судя по дампу последнему.
Российские E1 шлюзы Alvis. Модернизация УПАТС с E1, настройка Asterisk/FreePBX, подключение CRM
ded
Сообщения: 15689
Зарегистрирован: 26 авг 2010, 19:00

Re: Нет голоса у внешних клиентов

Сообщение ded »

Судя по дампу он и регистриируется успешно -

Код: Выделить всё

<--- SIP read from UDP:80.64.21.250:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP *.*.97.133:5060;branch=z9hG4bK5b2b0734;rport=58708;received=*.*.97.133
From: <sip:230333@80.64.21.250>;tag=as73c65a39
To: <sip:230333@80.64.21.250>;tag=254598dec69c11e6a3c62c44fd933994
Call-ID: 2733c4bd2fd69cab47a95120663245fc@127.0.0.1
CSeq: 103 REGISTER
Либо софтсвич провайдера раздражается на бессмысленные OPTIONS, либо НАТообразующее устройство начинает тупить.
Ответить
© 2008 — 2025 Asterisk.ru
Digium, Asterisk and AsteriskNOW are registered trademarks of Digium, Inc.
Design and development by PostMet-Netzwerk GmbH