приветствую!

компилю Asterisk-1.0.8
channels/h323 & openh323_1.12.2 & pwlib_1.5.2

звоню с н323 на сип телефон или обратно - тишина, никто не кого не слышет.

НАТа нет, все в одной сети.

завасит ли от того что на сервере где крутится * нет звуковой платы?

от звуковой платы независит совсем
а вот какая у тебя ОС и не было ли ошибок или много варнингов при сборке ?
какие кодеки при установке канала SIP-Oh323?

Так запустился сам модуль h323 ?
Как передаёшь звонок на H323 в extensions.conf?
Есть ли гейткипер?
Приведи лог звонка, такого вида -
-- Executing Macro("H323/ip$192.168.1.10:41861/26646", "stdexten|727|SIP/727") in new stack
-- Executing Dial("H323/ip$192.168.1.10:41861/26646", "SIP/727|20|Ttr") in new stack
-- Called 727
-- SIP/727-702b is ringing

модуль собрался
Warning не было

сейчас перекомпилил еще раз астериск на другой машине - опять тишина

схема такова

sip phone -- asterisk -- gnugk -- zebratelecom
192.168.0.100 192.168.0.41 192.168.0.40|82.208.67.xx 213.145.43.45

если звонить sip-h323 внутри астериска - все ок
если тупо сказать в Extensions.conf
exten=>_7.,1Dial(h323/${EXTEN}@213.145.43.45) = тоже все ок

вот лог с *
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
== Parsing '/etc/asterisk/enum.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/enum.conf': Found
== Parsing '/etc/asterisk/rtp.conf': Found
== RTP Allocating from port range 10000 -> 20000
Asterisk Ready.
*CLI> h.323 debug
H323 debug enabled
*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:
INVITE sip:78313253897@192.168.0.41;user=phone SIP/2.0
Call-ID:23bd4-244e8-ec6acc@192.168.0.100
From:100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To:<sip:78313253897@192.168.0.41;user=phone>
CSeq:100 INVITE
Via:SIP/2.0/UDP 192.168.0.100
Contact:100<sip:100@192.168.0.100;user=phone>
Content-Type:application/sdp
Content-Length:113

v=0
o=100 0 0 IN IP4 192.168.0.100
s=-
c=IN IP4 192.168.0.100
t=0 0
m=audio 8002 RTP/AVP 4 0 8
a=ptime:30

9 headers, 7 lines
Using latest request as basis request
Sending to 192.168.0.100 : 5060 (NAT)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.100;received=192.168.0.100;rport=5060
From: 100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To: <sip:78313253897@192.168.0.41;user=phone>;tag=as4eb99e30
Call-ID: 23bd4-244e8-ec6acc@192.168.0.100
CSeq: 100 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:78313253897@192.168.0.41>
Proxy-Authenticate: Digest realm="asterisk", nonce="297f598a"
Content-Length: 0


to 192.168.0.100:5060
Scheduling destruction of call '23bd4-244e8-ec6acc@192.168.0.100' in 15000 ms
Found user '100'


Sip read:
ACK sip:78313253897@192.168.0.41;user=phone SIP/2.0
Call-ID:23bd4-244e8-ec6acc@192.168.0.100
From:100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To:<sip:78313253897@192.168.0.41;user=phone>
CSeq:100 ACK
Via:SIP/2.0/UDP 192.168.0.100
Contact:100<sip:100@192.168.0.100;user=phone>
Content-Length:0


8 headers, 0 lines


Sip read:
INVITE sip:78313253897@192.168.0.41;user=phone SIP/2.0
Call-ID:23bd4-244e8-ec6acc@192.168.0.100
From:100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To:<sip:78313253897@192.168.0.41;user=phone>
CSeq:101 INVITE
Via:SIP/2.0/UDP 192.168.0.100
Contact:100<sip:100@192.168.0.100;user=phone>
PROXY-Authorization:Digest username="100",realm="asterisk",nonce="297f598a",uri="sip:78313253897@192.168.0.41;user=phone",response="073e13c48918e464b6f6085eb285c239"
Content-Type:application/sdp
Content-Length:113

v=0
o=100 0 0 IN IP4 192.168.0.100
s=-
c=IN IP4 192.168.0.100
t=0 0
m=audio 8002 RTP/AVP 4 0 8
a=ptime:30

10 headers, 7 lines
Using latest request as basis request
Sending to 192.168.0.100 : 5060 (NAT)
Found user '100'
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 192.168.0.100:8002
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 78313253897 in default
list_route: hop: <sip:100@192.168.0.100;user=phone>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100;received=192.168.0.100;rport=5060
From: 100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To: <sip:78313253897@192.168.0.41;user=phone>
Call-ID: 23bd4-244e8-ec6acc@192.168.0.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:78313253897@192.168.0.41>
Content-Length: 0


to 192.168.0.100:5060
-- Executing Dial("SIP/100-b7d2", "H323/78313253897") in new stack
Allowed Codecs:
Table:
G.711-uLaw-64k{sw} <1>
Set:
0:
0:
G.711-uLaw-64k{sw} <1>

-- Making call to 78313253897 using gatekeeper.
== New H.323 Connection created.
-- 100 is calling host 78313253897
-- Call token is ip$localhost/28273
-- Call reference is 28273
-- Called 78313253897
-- Sending SETUP message
=-= In OnAlerting for call 28273: sessionId=0
--- no logical channels
-- Ringing phone for "192.168.0.40"
-- H323/78313253897 is ringing
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.100;received=192.168.0.100;rport=5060
From: 100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To: <sip:78313253897@192.168.0.41;user=phone>;tag=as00602a67
Call-ID: 23bd4-244e8-ec6acc@192.168.0.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:78313253897@192.168.0.41>
Content-Length: 0


to 192.168.0.100:5060
=-= In OnConnectionEstablished for call 28273
-- Connection Established with "192.168.0.40"
-- H323/78313253897 answered SIP/100-b7d2
We're at 192.168.0.41 port 13650
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100;received=192.168.0.100;rport=5060
From: 100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To: <sip:78313253897@192.168.0.41;user=phone>;tag=as00602a67
Call-ID: 23bd4-244e8-ec6acc@192.168.0.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:78313253897@192.168.0.41>
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 4446 4446 IN IP4 192.168.0.41
s=session
c=IN IP4 192.168.0.41
t=0 0
m=audio 13650 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

to 192.168.0.100:5060


Sip read:
ACK sip:78313253897@192.168.0.41;user=phone SIP/2.0
Call-ID:23bd4-244e8-ec6acc@192.168.0.100
From:100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To:<sip:78313253897@192.168.0.41;user=phone>;tag=as00602a67
CSeq:101 ACK
Via:SIP/2.0/UDP 192.168.0.100
Contact:100<sip:100@192.168.0.100;user=phone>
Content-Length:0


8 headers, 0 lines


Sip read:
BYE sip:78313253897@192.168.0.41;user=phone SIP/2.0
Call-ID:23bd4-244e8-ec6acc@192.168.0.100
From:100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To:<sip:78313253897@192.168.0.41;user=phone>;tag=as00602a67
CSeq:102 BYE
Via:SIP/2.0/UDP 192.168.0.100
Contact:100<sip:100@192.168.0.100;user=phone>
Content-Length:0


8 headers, 0 lines
Sending to 192.168.0.100 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100;received=192.168.0.100;rport=5060
From: 100<sip:100@192.168.0.41;user=phone>;tag=ee2e3a
To: <sip:78313253897@192.168.0.41;user=phone>;tag=as00602a67
Call-ID: 23bd4-244e8-ec6acc@192.168.0.100
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:78313253897@192.168.0.41>
Content-Length: 0


to 192.168.0.100:5060
-- ClearCall: Request to clear call with token ip$localhost/28273
-- Sending RELEASE COMPLETE
== Spawn extension (default, 78313253897, 1) exited non-zero on 'SIP/100-b7d2'
-- Call with 192.168.0.40 completed (EndedByLocalUser)
== H.323 Connection deleted.
Destroying call '23bd4-244e8-ec6acc@192.168.0.100'

*CLI> stop
No such command 'stop' (type 'help' for help)
*CLI> stop now

Suse 9.1

linux:~ # uname -a
Linux linux 2.6.4-52-default #1 Wed Apr 7 02:08:30 UTC 2004 i686 i686 i386 GNU/Linux

Судя по схеме
sip phone -- asterisk -- gnugk -- zebratelecom
192.168.0.100 192.168.0.41 192.168.0.40|82.208.67.xx 213.145.43.45

asterisk У тебя работает на приватных IP.

А так как gnugk указвает только направление звонка (он же не h323-proxy),
то соединение устанавливается между 192.168.0.41 (наверное где-то же есть и NAT)
и 213.145.43.45.
Возможно поэтому и ничего не слышно.

спасибо за ответ!

gnugk поддерживает проксирование! проверялось с помощью н323 клиентов как аппаратных так и софтовых из левых подсеток.

2. проверялась также работа если на сервере (192.168.0.40|82.208.67.xx) крутятся сразу и астериск и gnugk
опять тишиа
http://www.asterisk-support.ru/Members/paganelp/AsteriskPlusgnugk/view?searchterm=1730

может есть смысл запустить tcpdump на gatekeeper'е ? хоть посмотреть какие там пакеты
и куда ходят ?