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Asterisk 1.6 и FreeBSD 8.0

Сообщений: 6521

Re: Asterisk 1.6 и FreeBSD 8.0

Скорее всего проблема в пакетизации.
Установите на Алкатели
g711_20
а на Астериске
allow=alaw:20
allow=ulaw:20
2010-05-19 17:33

Откуда: Саратов
Сообщений: 26

Re: Asterisk 1.6 и FreeBSD 8.0

Та же беда. Лог:
ssh*CLI>
-- Executing [241@office:1] Dial("SIP/281-00000002", "H323/241@212.33.19.195") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Making call to 241@212.33.19.195:1720 without gatekeeper.
ssh*CLI>== New H.323 Connection created.
ssh*CLI>-- root is calling host 241@212.33.19.195:1720
-- Call token is ip$localhost/17483
ssh*CLI>-- Call reference is 17483
-- DTMF Payload is [pt=101]
-- Called 241@212.33.19.195
Setting capabilities to 0x8 (alaw)
Capabilities in preference order is (alaw)
Allowed Codecs:
Table:
G.711-ALaw-64k <1>
UserInput/hookflash <2>
UserInput/RFC2833 <3>
UserInput/dtmf <4>
Set:
0:
0:
G.711-ALaw-64k <1>
1:
UserInput/hookflash <2>
2:
UserInput/RFC2833 <3>
UserInput/dtmf <4>

ssh*CLI>-- Sending SETUP message
ssh*CLI>=-= In OnAlerting for call 17483: sessionId=0
ssh*CLI>-- Ringing phone for " Denisov"
- Progress Indicator: 0
-- H323/212.33.19.195-3 is ringing
-- H323/212.33.19.195-3 is ringing
Using 212.33.19.196 for outbound H.245 transport
Peer capability is G.711-ALaw-64k <1>
Found peer capability G.711-ALaw-64k <1>, Asterisk code is 8, frame size (in ms) is 20
Peer capability is G.711-uLaw-64k <2>
Found peer capability G.711-uLaw-64k <2>, Asterisk code is 4, frame size (in ms) is 30
Peer capability is UserInput/dtmf <5>
Peer capabilities = 0xc (ulaw|alaw), ordered list is (alaw|ulaw)
ssh*CLI>-- Started logical channel: sending G.711-ALaw-64k
ssh*CLI> -- channelsOpen = 1
ssh*CLI>=-= In OnConnectionEstablished for call 17483
ssh*CLI> -- Connection Established with " Denisov"
-- H323/212.33.19.195-3 answered SIP/281-00000002
ssh*CLI>MyH323_ExternalRTPChannel::OnReceivedAckPDU
ssh*CLI> -- remoteIpAddress: 212.33.19.195
-- remotePort: 32082
External RTP Session Starting
RTP channel id 1 parameters:
ssh*CLI> -- remoteIpAddress: 212.33.19.195
-- remotePort: 32082
-- ExternalIpAddress: 212.33.19.196
-- ExternalPort: 13776
ssh*CLI>-- Started logical channel: receiving G.711-ALaw-64k
ssh*CLI> -- channelsOpen = 2
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 212.33.19.195
-- remotePort: 32080
-- ExternalIpAddress: 212.33.19.196
ssh*CLI> -- ExternalPort: 13776
ssh*CLI> channelsOpen = 1
ssh*CLI>ExternalRTPChannel Destroyed
ssh*CLI>-- Closing logical channel...
ssh*CLI> channelsOpen = 0
ssh*CLI>ExternalRTPChannel Destroyed
ssh*CLI>-- ClearCall: Request to clear call with token ip$localhost/17483, cause EndedByRemoteUser
ssh*CLI>-- Sending RELEASE COMPLETE
ssh*CLI>-- ClearCall: Request to clear call with token ip$localhost/17483, cause EndedByTransportFail
-- Denisov has cleared the call
ssh*CLI>== H.323 Connection deleted.
== Spawn extension (office, 241, 1) exited non-zero on 'SIP/281-00000002'
ssh*CLI>
2010-05-19 17:51

Сообщений: 6521

Re: Asterisk 1.6 и FreeBSD 8.0

Ну а разность пактизации вот тут
Found peer capability G.711-ALaw-64k <1>, Asterisk code is 8, frame size (in ms) is 20
Peer capability is G.711-uLaw-64k <2>
Found peer capability G.711-uLaw-64k <2>, Asterisk code is 4, frame size (in ms) is 30
Peer capability is UserInput/dtmf <5>
разве не видна невооружённым глазом?

Вы всё таки определитесь с каналом и конфигами.
Или NuFone chan_h323.so или
OOH323 Objective Systems H323 Channel Driver - chan_ooh323.so

NuFone - h323.conf синтаксис такой
[Cisco1760]
type=peer
host=10.198.39.243
port=1720
context=from-trunk

а Objective Systems - ooh323.conf такой
[Cisco1760]
type=friend
ip=10.198.39.243
port=1720 ; UPDATE with appropriate port
rtptimeout=60
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=alaw

CLI>core set debug 5
даст высойкий уровень дебага, результат смотреть в
$less /var/log/asterisk/full | grep chan_h323.c
2010-05-19 17:55

Откуда: Саратов
Сообщений: 26

Re: Asterisk 1.6 и FreeBSD 8.0

ssh*CLI> core set debug 5
Core debug was 0 and is now 5
ssh*CLI> h323 set debug
H.323 debug enabled
-- Executing [241@office:1] Dial("SIP/281-00000000", "H323/241@alcatel") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Making call to 241@212.33.19.195:1720 without gatekeeper.
ssh*CLI>== New H.323 Connection created.
ssh*CLI>-- root is calling host 241@212.33.19.195:1720
-- Call token is ip$localhost/18543
-- Call reference is 18543
ssh*CLI>-- DTMF Payload is [pt=101]
-- Called 241@alcatel
Setting capabilities to 0xc (ulaw|alaw)
Capabilities in preference order is (alaw|ulaw)
Allowed Codecs:
ssh*CLI> Table:
G.711-ALaw-64k <1>
G.711-uLaw-64k <2>
UserInput/hookflash <3>
UserInput/RFC2833 <4>
UserInput/dtmf <5>
Set:LI>
0:LI>
0:>
G.711-ALaw-64k <1>
G.711-uLaw-64k <2>
1:>
UserInput/hookflash <3>
2:>
UserInput/RFC2833 <4>
UserInput/dtmf <5>
ssh*CLI>
ssh*CLI>-- Sending SETUP message
ssh*CLI>=-= In OnAlerting for call 18543: sessionId=0
ssh*CLI>-- Ringing phone for " Denisov"
ssh*CLI> - Progress Indicator: 0
-- H323/alcatel-1 is ringing
-- H323/alcatel-1 is ringing
Using 212.33.19.196 for outbound H.245 transport
Peer capability is G.711-ALaw-64k <1>
Found peer capability G.711-ALaw-64k <1>, Asterisk code is 8, frame size (in ms) is 20
Peer capability is G.711-uLaw-64k <2>
Found peer capability G.711-uLaw-64k <2>, Asterisk code is 4, frame size (in ms) is 20
Peer capability is UserInput/dtmf <5>
Peer capabilities = 0xc (ulaw|alaw), ordered list is (alaw|ulaw)
ssh*CLI>-- Started logical channel: sending G.711-ALaw-64k
ssh*CLI> -- channelsOpen = 1
ssh*CLI>=-= In OnConnectionEstablished for call 18543
ssh*CLI> -- Connection Established with " Denisov"
-- H323/alcatel-1 answered SIP/281-00000000
ssh*CLI>MyH323_ExternalRTPChannel::OnReceivedAckPDU
ssh*CLI> -- remoteIpAddress: 212.33.19.195
-- remotePort: 32162
ssh*CLI> External RTP Session Starting
ssh*CLI> RTP channel id 1 parameters:
ssh*CLI> -- remoteIpAddress: 212.33.19.195
-- remotePort: 32162
ssh*CLI> -- ExternalIpAddress: 212.33.19.196
ssh*CLI> -- ExternalPort: 17780
ssh*CLI>-- Started logical channel: receiving G.711-ALaw-64k
ssh*CLI> -- channelsOpen = 2
ssh*CLI> External RTP Session Starting
RTP channel id 1 parameters:
ssh*CLI> -- remoteIpAddress: 212.33.19.195
ssh*CLI> -- remotePort: 32160
ssh*CLI> -- ExternalIpAddress: 212.33.19.196
ssh*CLI> -- ExternalPort: 17780
ssh*CLI> channelsOpen = 1
ssh*CLI>ExternalRTPChannel Destroyed
ssh*CLI>-- Closing logical channel...
ssh*CLI> channelsOpen = 0
ssh*CLI>ExternalRTPChannel Destroyed
ssh*CLI>-- ClearCall: Request to clear call with token ip$localhost/18543, cause EndedByRemoteUser
ssh*CLI>-- Sending RELEASE COMPLETE
ssh*CLI>-- ClearCall: Request to clear call with token ip$localhost/18543, cause EndedByTransportFail
-- Denisov has cleared the call
ssh*CLI>== H.323 Connection deleted.
== Spawn extension (office, 241, 1) exited non-zero on 'SIP/281-00000000'
Really destroying SIP dialog '31DF8D65A896440DB9B100484255E5170xc0a801bd' Method: ACK
ssh*CLI> exit

Размер кадров стал норм. Не обратили внимания после какого действия абонент алкатели начал слышать сам себя. Абонент састера продолжает слышать абонента алкатели.

cat /var/log/asterisk/full | grep h323
[May 20 09:24:34] DEBUG[85975] chan_h323.c: type=H323, format=8, data=241@alcatel.
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Extension: 241 Host: alcatel
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Calling to 241@alcatel on H323/alcatel-3
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Placing outgoing call to 241@212.33.19.195:1720, 101
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Setting capabilities for connection ip$localhost/18545
[May 20 09:24:34] DEBUG[85975] chan_h323.c: local prefs[0]=alaw:20
[May 20 09:24:34] DEBUG[85975] chan_h323.c: local prefs[1]=ulaw:20
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Capabilities for connection ip$localhost/18545 is set
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Received ALERT/PROGRESS message for self-generated tones
[May 20 09:24:34] DEBUG[85975] chan_h323.c: Ringing on ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Got remote capabilities from connection ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: prefs[0]=alaw:20
[May 20 09:24:36] DEBUG[85975] chan_h323.c: prefs[1]=ulaw:20
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Created RTP channel
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Setting NAT on RTP to 0
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Sending RTP 'US' 212.33.19.196:19442
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Call ip$localhost/18545 answered
[May 20 09:24:36] DEBUG[85975] chan_h323.c: OH323: Indicating 20 on ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: OH323: Indicated 20 on ip$localhost/18545, res=0
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Setting up RTP connection for ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Native format is set to 8 from 8 by RTP payload type 8
[May 20 09:24:36] DEBUG[85975] chan_h323.c: RTP connection prepared for ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Setting up RTP connection for ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Native format is set to 8 from 8 by RTP payload type 8
[May 20 09:24:36] DEBUG[85975] chan_h323.c: RTP connection prepared for ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Sending RTP 'US' 212.33.19.196:19442
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Setting up RTP connection for ip$localhost/18545
[May 20 09:24:36] DEBUG[85975] chan_h323.c: Native format is set to 8 from 8 by RTP payload type 8
[May 20 09:24:36] DEBUG[85975] chan_h323.c: RTP connection prepared for ip$localhost/18545
[May 20 09:24:40] DEBUG[85975] chan_h323.c: Cleaning connection to ip$localhost/18545
[May 20 09:24:40] DEBUG[85975] chan_h323.c: Connection to ip$localhost/18545 cleaned
[May 20 09:24:40] DEBUG[85975] chan_h323.c: Hanging up and scheduling destroy of call H323/alcatel-3



последний h323.conf


ssh# cat h323.conf | grep -v ';'
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw:20
allow=ulaw:20
h245tunneling=no
fastStart=no
dtmfmode=rfc2833

AcceptAnonymous = yes
context=office

[alcatel]
type=peer
context=office
host=212.33.19.195
port=1720
disallow=all
allow=alaw:20
allow=ulaw:20
h245tunneling=no
fastStart=no
canreinvite=no

ssh#

2010-05-20 10:25

Сообщений: 6521

Re: Asterisk 1.6 и FreeBSD 8.0

Дебаг RTP правильный до безобразия.
Смените софтфон.
2010-05-20 10:33

Откуда: Саратов
Сообщений: 26

Re: Asterisk 1.6 и FreeBSD 8.0

Софтфон сменили, результата нет))))
2010-05-21 09:55

Откуда: Саратов
Сообщений: 26

Re: Asterisk 1.6 и FreeBSD 8.0

А дебаг такой красивый, потому что мы старались, печатали его сами))))

Шутка.
2010-05-21 10:16

Сообщений: 1129

Re: Asterisk 1.6 и FreeBSD 8.0

тогда окупируйтесь в талмудах о двух протоколах H323,SIP
вооружившись дебагами и дампом tcpdump + варешарком
ищите проблему
ортодоксальный антиастерискер || антилинуксоид! (астериск || линукс) - иррациональное решение!. и здесь я тоже http://forum.asterisk.ru
2010-05-21 20:21

Откуда: Саратов
Сообщений: 26

Re: Asterisk 1.6 и FreeBSD 8.0

Может ошибка какая то тупая, по моему не знанию? Давайте выложу остальные конфиги?
2010-05-26 11:24

Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Asterisk 1.6 и FreeBSD 8.0

в талмудах искать лень и от вайршарка страшно?

;)
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2010-05-26 11:28

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