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спросил 2018-03-27 23:52:46 +0400

ramadan Gravatar ramadan

asteriks 11 включаю TLS, но порт 5061 не слушает

Доброго времени суток. Имеется asterisk 11.14.0

module show like res_srtp.so
Module                         Description                              Use Count
res_srtp.so                    Secure RTP (SRTP)                        0
1 modules loaded

Добавил в sip.conf

[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/certificate/asterisk.pem ; путь к локальному сертификату
tlscafile=/etc/asterisk/certificate/ca.crt ; путь к доверенному центру сертификации
tlscipher=ALL
tlsclientmethod=tlsv1

Сгенерил сертификаты с помощью ./asttlscert

Перезапустил астериск, но он 5061 порт так и не начал слушать

netstat -lanp | grep asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           2454/asterisk
unix  2      [ ACC ]     STREAM     LISTENING     26203    2454/asterisk        /run/asterisk/asterisk.ctl
unix  3      [ ]         STREAM     CONNECTED     28564    2454/asterisk

Ну и понятное дело, что в итоге так как на порту никто ничего не слушает:

openssl s_client -connect 127.0.0.1:5061
socket: Connection refused
connect:errno=111

Подскажите пожалуйста в чем может быть проблема.

asteriks 11 включаю TLS, но порт 5061 не слушает

Доброго времени суток. Имеется asterisk 11.14.0

module show like res_srtp.so
Module                         Description                              Use Count
res_srtp.so                    Secure RTP (SRTP)                        0
1 modules loaded

Добавил в sip.conf

[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/certificate/asterisk.pem ; путь к локальному сертификату
tlscafile=/etc/asterisk/certificate/ca.crt ; путь к доверенному центру сертификации
tlscipher=ALL
tlsclientmethod=tlsv1

Сгенерил сертификаты с помощью ./asttlscert

Перезапустил астериск, но он 5061 порт так и не начал слушать

netstat -lanp | grep asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           2454/asterisk
unix  2      [ ACC ]     STREAM     LISTENING     26203    2454/asterisk        /run/asterisk/asterisk.ctl
unix  3      [ ]         STREAM     CONNECTED     28564    2454/asterisk

Ну и понятное дело, что в итоге так как на порту никто ничего не слушает:

openssl s_client -connect 127.0.0.1:5061
socket: Connection refused
connect:errno=111

Подскажите пожалуйста в чем может быть проблема.

asteriks 11 включаю TLS, но порт 5061 не слушает

Доброго времени суток. Имеется asterisk 11.14.0

module show like res_srtp.so
Module                         Description                              Use Count
res_srtp.so                    Secure RTP (SRTP)                        0
1 modules loaded

Добавил в sip.conf

[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/certificate/asterisk.pem ; путь к локальному сертификату
tlscafile=/etc/asterisk/certificate/ca.crt ; путь к доверенному центру сертификации
tlscipher=ALL
tlsclientmethod=tlsv1

Сгенерил сертификаты с помощью ./asttlscert

Перезапустил астериск, но он 5061 порт так и не начал слушать

netstat -lanp | grep asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           2454/asterisk
unix  2      [ ACC ]     STREAM     LISTENING     26203    2454/asterisk        /run/asterisk/asterisk.ctl
unix  3      [ ]         STREAM     CONNECTED     28564    2454/asterisk

Ну и понятное дело, что в итоге так как на порту никто ничего не слушает:

openssl s_client -connect 127.0.0.1:5061
socket: Connection refused
connect:errno=111

Подскажите пожалуйста в чем может быть проблема.

вот лог при старте

[Mar 28 09:48:03] Asterisk 11.14.1 built by mockbuild @ buildhw-11.phx2.fedoraproject.org on a x86_64 running Linux on 2014-11-22 01:36:05 UTC
[Mar 28 09:48:03] NOTICE[19260] cdr.c: CDR simple logging enabled.
[Mar 28 09:48:03] NOTICE[19260] loader.c: 185 modules will be loaded.
[Mar 28 09:48:03] NOTICE[19260] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Mar 28 09:48:03] WARNING[19260] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Mar 28 09:48:04] VERBOSE[19260] chan_sip.c: SIP channel loading...
[Mar 28 09:48:04] VERBOSE[19260] tcptls.c: SSL certificate ok
[Mar 28 09:48:04] WARNING[19260] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the  global setting can make
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! will be sent to a different port than replies for an existing peer/user. If at all possible,
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user.
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! (config category='siplife' global force_rport='No' peer/user force_rport='Yes')
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the  global setting can make
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! will be sent to a different port than replies for an existing peer/user. If at all possible,
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user.
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! (config category='gsmgw1' global force_rport='No' peer/user force_rport='Yes')
[Mar 28 09:48:04] NOTICE[19260] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'default' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'public' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: Starting AEL load process.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 190-205: The macro num_norm does not end with a return; I will insert one.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 207-216: The macro record_call does not end with a return; I will insert one.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 218-226: The macro record_out does not end with a return; I will insert one.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'public' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'default' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] VERBOSE[19260] asterisk.c: Asterisk Ready.
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '501' is now Reachable. (12ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '502' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '504' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '503' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '506' is now Reachable. (7ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '505' is now Reachable. (11ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '507' is now Reachable. (52ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '508' is now Reachable. (52ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '511' is now Reachable. (15ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '509' is now Reachable. (37ms / 2000ms)

asteriks 11 включаю TLS, но порт 5061 не слушает

Доброго времени суток. Имеется asterisk 11.14.0

module show like res_srtp.so
Module                         Description                              Use Count
res_srtp.so                    Secure RTP (SRTP)                        0
1 modules loaded

Добавил в sip.conf

[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/certificate/asterisk.pem ; путь к локальному сертификату
tlscafile=/etc/asterisk/certificate/ca.crt ; путь к доверенному центру сертификации
tlscipher=ALL
tlsclientmethod=tlsv1

Сгенерил сертификаты с помощью ./asttlscert

Перезапустил астериск, но он 5061 порт так и не начал слушать

netstat -lanp | grep asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           2454/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           2454/asterisk
unix  2      [ ACC ]     STREAM     LISTENING     26203    2454/asterisk        /run/asterisk/asterisk.ctl
unix  3      [ ]         STREAM     CONNECTED     28564    2454/asterisk

Ну и понятное дело, что в итоге так как на порту никто ничего не слушает:

openssl s_client -connect 127.0.0.1:5061
socket: Connection refused
connect:errno=111

Подскажите пожалуйста в чем может быть проблема.

вот лог при старте

[Mar 28 09:48:03] Asterisk 11.14.1 built by mockbuild @ buildhw-11.phx2.fedoraproject.org on a x86_64 running Linux on 2014-11-22 01:36:05 UTC
[Mar 28 09:48:03] NOTICE[19260] cdr.c: CDR simple logging enabled.
[Mar 28 09:48:03] NOTICE[19260] loader.c: 185 modules will be loaded.
[Mar 28 09:48:03] NOTICE[19260] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Mar 28 09:48:03] WARNING[19260] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Mar 28 09:48:04] VERBOSE[19260] chan_sip.c: SIP channel loading...
[Mar 28 09:48:04] VERBOSE[19260] tcptls.c: SSL certificate ok
[Mar 28 09:48:04] WARNING[19260] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the  global setting can make
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! will be sent to a different port than replies for an existing peer/user. If at all possible,
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user.
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! (config category='siplife' global force_rport='No' peer/user force_rport='Yes')
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the  global setting can make
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! will be sent to a different port than replies for an existing peer/user. If at all possible,
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! use the global 'nat' setting and do not set 'nat' per peer/user.
[Mar 28 09:48:04] WARNING[19260] chan_sip.c: !!! (config category='gsmgw1' global force_rport='No' peer/user force_rport='Yes')
[Mar 28 09:48:04] NOTICE[19260] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'default' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'public' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: Starting AEL load process.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 190-205: The macro num_norm does not end with a return; I will insert one.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 207-216: The macro record_call does not end with a return; I will insert one.
[Mar 28 09:48:04] WARNING[19260] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 218-226: The macro record_out does not end with a return; I will insert one.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'public' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] WARNING[19260] pbx.c: Context 'default' tries to include nonexistent context 'demo'
[Mar 28 09:48:04] NOTICE[19260] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Mar 28 09:48:04] VERBOSE[19260] asterisk.c: Asterisk Ready.
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '501' is now Reachable. (12ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '502' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '504' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '503' is now Reachable. (18ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '506' is now Reachable. (7ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '505' is now Reachable. (11ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '507' is now Reachable. (52ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '508' is now Reachable. (52ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '511' is now Reachable. (15ms / 2000ms)
[Mar 28 09:48:04] NOTICE[19290] chan_sip.c: Peer '509' is now Reachable. (37ms / 2000ms)

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