Нет входящего звука с аппаратов Yealink w60p?

Question

[daler86]

Здравствуйте!
Помогите пожалуйста с решением проблемы,

В сети имеется asterisk (freepbx) и сеть филиала, как на схеме. Астериск 2 сетевые карты, один для внутренней сети другой для подключения к провайдеру. Недавно приобрели новые аппараты Yealink. Когда с данных с этих аппаратов звонишь наружу (мобильные операторы) тебя слышно, но ты не слышешь. Таких проблем на аппаратах Grandtream нету.

Comments

[roswell]

Я бы попробовал включить 'Enable Outbound Proxy Server', в поле 'Outbound Proxy 1' указать адрес VoIP-сервера, и включить 'NAT'.

[Valentin Barbolin]

Покажи вывод команд с астериска.

asterisk -rx "sip show settings"

asterisk -rx "sip show peer 777"

[daler86]

Andrey Barbolin,
asterisk -rx "sip show settings"

Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5068
TCP SIP Bindaddress: 0.0.0.0:5068
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-13.0.192.16(13.7.1)
SDP Session Name: Asterisk PBX 13.7.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:
---------------------------
SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: 15.12.55.5:0
Externrefresh: 10
Localnet: 192.168.40.0/255.255.255.0


Global Signalling Settings:
---------------------------
Codecs: (ulaw|alaw|h264|mpeg4)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:
-----------------
Allowed transports: TCP
Outbound transport: TCP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: ru
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
----
asterisk -rx "sip show peer 777"
Peer 777 not found.

[daler86]

roswell,
Enable Outbound Proxy Server поставиль ip сервер FreePBX не помогло,
и включиль 'NAT' такая же проблема .

[Valentin Barbolin]

daler86, Судя по выводу команд, у вас используется PJSIP, соответственно надо
asterisk -rx "pjsip show settings"

asterisk -rx "pjsip show endpoint 777"

[daler86]

Andrey Barbolin,
asterisk -rx "pjsip show settings"

Global Settings:

ParameterName : ParameterValue
====================================================
debug : no
default_from_user : asterisk
default_outbound_endpoint : dpma_endpoint
endpoint_identifier_order : ip,username,anonymous
keep_alive_interval : 0
max_forwards : 70
max_initial_qualify_time : 0
user_agent : FPBX-13.0.192.16(13.7.1)

System Settings:

ParameterName : ParameterValue
==========================================
compact_headers : false
disable_tcp_switch : true
threadpool_auto_increment : 5
threadpool_idle_timeout : 60
threadpool_initial_size : 0
threadpool_max_size : 50
timer_b : 32000
timer_t1 : 500
----------------------------------

asterisk -rx "pjsip show endpoint 777"

Endpoint:
I/OAuth:
Aor:
Contact:
Transport:
Identify:
Match:
Channel:
Exten: CLCID:
=========================================================================================

Endpoint: 777/777 Not in use 0 of inf
InAuth: 777-auth/777
Aor: 777 1
Contact: 777/sip:777@192.168.40.30:5060 3e409db323 Avail 5.890
Identify: 777-identify/777

ParameterName : ParameterValue
====================================================
100rel : yes
accountcode :
aggregate_mwi : true
allow : (ulaw|alaw)
allow_subscribe : true
allow_transfer : true
aors : 777
auth : 777-auth
call_group :
callerid : "device" <777>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
context : from-internal
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : true
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username
inband_progress : false
language : ru
mailboxes :
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_suggest : default
mwi_from_user :
named_call_group :
named_pickup_group :
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group :
record_off_feature : automixmon
record_on_feature : automixmon
rewrite_contact : true
rpid_immediate : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : true
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport :
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false

[Valentin Barbolin]

daler86, Выключили эту настройку у 777
direct_media : true

должно быть
direct_media : false

[daler86]

Andrey Barbolin,
ParameterName : ParameterValue
====================================================
100rel : yes
accountcode :
aggregate_mwi : true
allow : (ulaw|alaw)
allow_subscribe : true
allow_transfer : true
aors : 777
auth : 777-auth
call_group :
callerid : "device" <777>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
context : from-internal
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username
inband_progress : false
language : ru
mailboxes :
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_suggest : default
mwi_from_user :
named_call_group :
named_pickup_group :
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group :
record_off_feature : automixmon
record_on_feature : automixmon
rewrite_contact : true
rpid_immediate : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : true
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport :
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false

Не помогло

[Valentin Barbolin]

Можно дамп посмотреть?
ETH_LOCAL - интерфейст сервера в локальной сети
PHONE_IP - IP адрес проблемного телефона

Запустить tcpdump - ni ETH_LOCAL host PHONE_IP -w call.pcap, сделать звонок, потом Ctrl+c, и сюда выложить файлик call.pcap

[daler86]

Andrey Barbolin,

https://drive.google.com/file/d/1iWHCBlNA9fSqTNSod...

[Valentin Barbolin]

Выключи на телефоне кодек 711u

[daler86]

Andrey Barbolin,
Огромное спасибо! Заработало!

[Valentin Barbolin]

daler86, Можешь вернуть кодек на телефон, но поменять приоритет. Поставить 711а выше по списку.

Оформил как отдельный ответ.