1 | изначальная версия редактировать | |
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.
2 | No.2 Revision редактировать |
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.
3 | No.3 Revision редактировать |
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.