Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас
Задайте Ваш вопрос

История изменений [назад]

нажмите, чтобы скрыть/показать версии 1
изначальная версия
редактировать

спросил 2015-03-13 18:42:13 +0400

wellus Gravatar wellus

ooh323 panasonic TDA

Добрый день!
Настроил связку Asterisk с TDA600 и c TDE600 по h323
TDE600 с Asterisk работает
TDA600 с Asterisk работает, но с нюансами:

Звонок с Asterisk на TDA600 - работает без нареканий Звонок с TDA600 на Asterisk - работает ТОЛЬКО с запущенным "core set debug 1"...

Как "core set debug 1" - может влиять на работу ooh323 ?

Вот звонок с "core set debug off"

        -- Executing [s@macro-dial-one:43] Dial("OOH323/peer-26", "SIP/2209,,tr") in new stack
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Called SIP/2209
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
      == Begin MixMonitor Recording OOH323/peer-26
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
        -- SIP/2209-00000019 is ringing
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019145, ts 000160, len 000160)
...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019248, ts 016640, len 000160)
    ---   ooh323_update_writeformat alaw/20
    ---   find_call
    +++   find_call
    Writeformat before update slin/(alaw)
    +++   ooh323_update_writeformat
    ---   setup_rtp_connection a.b.c.d:12006
    ---   find_call
    +++   find_call
    +++   setup_rtp_connection
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019249, ts 016800, len 000160)
 ...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019318, ts 027840, len 000160)
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
        -- SIP/2209-00000019 answered OOH323/peer-26
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    --- ooh323_answer
    +++ ooh323_answer
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    ----- ooh323_indicate 20 on call ooh323c_25
    ++++  ooh323_indicate 20 on ooh323c_25 is -1
    ---   onCallEstablished ooh323c_25
    ---   find_call
    +++   find_call
    +++   onCallEstablished ooh323c_25
    ---   close_rtp_connection
    ---   find_call
    +++   find_call
    +++   close_rtp_connection
    ---   onCallCleared ooh323c_25 
    ---   find_call
    +++   find_call
    +++   onCallCleared
        -- Executing [h@macro-dial-one:1] Macro("OOH323/peer-26", "hangupcall,") in new stack

Звонок с "core set debug off" выкладывать не буду, кроме вот этого куска:

[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:6207 sip_call: Outgoing Call for 2209
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13085 add_sdp: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13086 add_sdp: ** Our prefcodec: (alaw) 
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method INVITE - callid 70dfc6341469065058bb67bb1b826731@192.168.1.12:5060
    -- Called SIP/2209
----- ooh323_indicate 3 on call ooh323c_31
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_ooh323.c:1307 ooh323_indicate: Sending manual ringback for ooh323c_31, res = 0
++++  ooh323_indicate 3 on ooh323c_31 is -1
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:4667 ast_indicate_data: Driver for channel 'OOH323/peer-32' does not support indication 3, emulating it
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:5405 set_format: Set channel OOH323/peer-32 to write format slin

Но звонки реально сразу проходят.

ooh323 panasonic TDA

Добрый день!
Настроил связку Asterisk с TDA600 и c TDE600 по h323

elastix*CLI> core show version 
Asterisk 11.13.0 built by palosanto @ rpmbuild64-2.elastix.palosanto.com on a x86_64 running Linux on 2014-10-04 01:12:50 UTC

TDE600 с Asterisk работает
TDA600 с Asterisk работает, но с нюансами:

Звонок с Asterisk на TDA600 - работает без нареканий Звонок с TDA600 на Asterisk - работает ТОЛЬКО с запущенным "core set debug 1"...

Как "core set debug 1" - может влиять на работу ooh323 ?

Вот звонок с "core set debug off"

        -- Executing [s@macro-dial-one:43] Dial("OOH323/peer-26", "SIP/2209,,tr") in new stack
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Called SIP/2209
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
      == Begin MixMonitor Recording OOH323/peer-26
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
        -- SIP/2209-00000019 is ringing
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019145, ts 000160, len 000160)
...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019248, ts 016640, len 000160)
    ---   ooh323_update_writeformat alaw/20
    ---   find_call
    +++   find_call
    Writeformat before update slin/(alaw)
    +++   ooh323_update_writeformat
    ---   setup_rtp_connection a.b.c.d:12006
    ---   find_call
    +++   find_call
    +++   setup_rtp_connection
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019249, ts 016800, len 000160)
 ...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019318, ts 027840, len 000160)
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
        -- SIP/2209-00000019 answered OOH323/peer-26
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    --- ooh323_answer
    +++ ooh323_answer
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    ----- ooh323_indicate 20 on call ooh323c_25
    ++++  ooh323_indicate 20 on ooh323c_25 is -1
    ---   onCallEstablished ooh323c_25
    ---   find_call
    +++   find_call
    +++   onCallEstablished ooh323c_25
    ---   close_rtp_connection
    ---   find_call
    +++   find_call
    +++   close_rtp_connection
    ---   onCallCleared ooh323c_25 
    ---   find_call
    +++   find_call
    +++   onCallCleared
        -- Executing [h@macro-dial-one:1] Macro("OOH323/peer-26", "hangupcall,") in new stack

Звонок с "core set debug off" выкладывать не буду, кроме вот этого куска:

[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:6207 sip_call: Outgoing Call for 2209
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13085 add_sdp: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13086 add_sdp: ** Our prefcodec: (alaw) 
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method INVITE - callid 70dfc6341469065058bb67bb1b826731@192.168.1.12:5060
    -- Called SIP/2209
----- ooh323_indicate 3 on call ooh323c_31
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_ooh323.c:1307 ooh323_indicate: Sending manual ringback for ooh323c_31, res = 0
++++  ooh323_indicate 3 on ooh323c_31 is -1
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:4667 ast_indicate_data: Driver for channel 'OOH323/peer-32' does not support indication 3, emulating it
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:5405 set_format: Set channel OOH323/peer-32 to write format slin

Но звонки реально сразу проходят.

ooh323 panasonic TDA

Добрый день!
Настроил связку Asterisk с TDA600 и c TDE600 по h323

elastix*CLI> core show version 
Asterisk 11.13.0 built by palosanto @ rpmbuild64-2.elastix.palosanto.com on a x86_64 running Linux on 2014-10-04 01:12:50 UTC

TDE600 с Asterisk работает
TDA600 с Asterisk работает, но с нюансами:

Звонок с Asterisk на TDA600 - работает без нареканий Звонок с TDA600 на Asterisk - работает ТОЛЬКО с запущенным "core set debug 1"...

Как "core set debug 1" - может влиять на работу ooh323 ?

Вот звонок с "core set debug off"

        -- Executing [s@macro-dial-one:43] Dial("OOH323/peer-26", "SIP/2209,,tr") in new stack
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Called SIP/2209
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
      == Begin MixMonitor Recording OOH323/peer-26
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
        -- SIP/2209-00000019 is ringing
    ----- ooh323_indicate 3 on call ooh323c_25
    ++++  ooh323_indicate 3 on ooh323c_25 is -1
    [2015-03-13 15:11:08] WARNING[22072][C-0000001c]: translate.c:343 framein: no samples for alawtolin
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019145, ts 000160, len 000160)
...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019248, ts 016640, len 000160)
    ---   ooh323_update_writeformat alaw/20
    ---   find_call
    +++   find_call
    Writeformat before update slin/(alaw)
    +++   ooh323_update_writeformat
    ---   setup_rtp_connection a.b.c.d:12006
    ---   find_call
    +++   find_call
    +++   setup_rtp_connection
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019249, ts 016800, len 000160)
 ...
    Sent RTP packet to      a.b.c.d:12006 (type 08, seq 019318, ts 027840, len 000160)
    ----- ooh323_indicate 33 on call ooh323c_25
    ++++  ooh323_indicate 33 on ooh323c_25 is -1
    ----- ooh323_indicate 22 on call ooh323c_25
    ++++  ooh323_indicate 22 on ooh323c_25 is -1
        -- SIP/2209-00000019 answered OOH323/peer-26
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    --- ooh323_answer
    +++ ooh323_answer
    ----- ooh323_indicate -1 on call ooh323c_25
    ++++  ooh323_indicate -1 on ooh323c_25 is -1
    ----- ooh323_indicate 20 on call ooh323c_25
    ++++  ooh323_indicate 20 on ooh323c_25 is -1
    ---   onCallEstablished ooh323c_25
    ---   find_call
    +++   find_call
    +++   onCallEstablished ooh323c_25
    ---   close_rtp_connection
    ---   find_call
    +++   find_call
    +++   close_rtp_connection
    ---   onCallCleared ooh323c_25 
    ---   find_call
    +++   find_call
    +++   onCallCleared
        -- Executing [h@macro-dial-one:1] Macro("OOH323/peer-26", "hangupcall,") in new stack

Звонок с "core set debug off" 1" выкладывать не буду, кроме вот этого куска:

[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:6207 sip_call: Outgoing Call for 2209
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13085 add_sdp: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:13086 add_sdp: ** Our prefcodec: (alaw) 
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method INVITE - callid 70dfc6341469065058bb67bb1b826731@192.168.1.12:5060
    -- Called SIP/2209
----- ooh323_indicate 3 on call ooh323c_31
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: chan_ooh323.c:1307 ooh323_indicate: Sending manual ringback for ooh323c_31, res = 0
++++  ooh323_indicate 3 on ooh323c_31 is -1
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:4667 ast_indicate_data: Driver for channel 'OOH323/peer-32' does not support indication 3, emulating it
[2015-03-13 15:33:38] DEBUG[22327][C-00000022]: channel.c:5405 set_format: Set channel OOH323/peer-32 to write format slin

Но звонки реально сразу проходят.

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