thr3ads.net - asterisk users - [Asterisk-Users] H323 one way audio using oh323 [Oct 2005]

If this information is useful, please help other people find it:
Share via:

mik sib

2005-Oct-31 03:05 UTC

[Asterisk-Users] H323 one way audio using oh323

Hi all,

through oh323 i can register to my gatekeeper and make
and receive calls.

My gatekeeper routes the incoming call as well as the
outgoing.

The problem is simply that i can't ear nothing from my
SIP ipPhones. I can ear my voice from a normal
telephone in my SIP phone but no viceversa.

How can i debug this situation ? I've no errors in the
log or at the asterisk startup.
How to understand what's happening ?
I've tryed different phones also.
any idea ?
thank you very much
Mik


Here's my oh323.conf
 Configuration of OpenH323 channel driver
------------------------------------------
Version: 0.7.3
Listening on address: 10.0.0.253:1720
Gatekeeper used: Nortel_H323_Gatekeeper@192.168.1.10
(Registered)
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref. order: ulaw<0>
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 10000 - 20000
UDP (RAS) port range: 10000 - 20000
UDP (RTP) port range: 10000 - 20000
IP Type-of-Service value: 0
User input mode: rfc2833
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
Max number of simultaneous H.323 calls: 100
Max call rate (ingress direction): 1.00/30
Default language: en
Default music class: default
Default context: voip-h323

doing a call with the ip phone to the outside world
through the gatekeeper

[2]WrapperAPI::h323_make_call: Making call.
[2]WrapH323EndPoint::MakeCall: Making call to
0258115040
[4]WrapH323EndPoint::CreateConnection: Creating a
H323Connection [32066]
[2]WrapH323Connection::WrapH323Connection: Creation of
WrapH323Connection based on user data.
[2]WrapH323Connection::WrapH323Connection: Call is
outgoing.
[4]WrapH323Connection::WrapH323Connection:
WrapH323Connection created.
[3]WrapH323EndPoint::MakeCall: Call token is
ip$localhost/32066
[3]WrapH323EndPoint::MakeCall: Call reference is 32066
[2]WrapH323Connection::OnSendSignalSetup: Sending
SETUP message...
[3]WrapH323Connection::OnSendSignalSetup: Setting
display name 0432281316 Fabio Violino
[3]WrapH323Connection::OnSendSignalSetup: Setting
calling party number test419
[2]WrapH323Connection::OnAlerting: Ringing phone for
"0258115040" ...
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
RECODER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=45)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 45,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for recording using 1x320 byte
buffers.
[3]WrapH323Connection::OnEstablished:
WrapH323Connection [ip$localhost/32066] established
(FastStartDisabled/noH245Tunneling)
[3]WrapH323EndPoint::OnConnectionEstablished:
Connection [ip$localhost/32066] established.
[3]WrapH323EndPoint::GetConnectionInfo:
[ip$localhost/32066] RTP Media:
10.0.0.253:10004-0.0.0.0:0
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
PLAYER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=43)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 43,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for playing using 1x320 byte
buffers.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[3]WrapH323EndPoint::SetClearCallCause: Setting the
Q.931 cause code of connection [ip$localhost/32066],
at 16
[2]WrapperAPI::h323_clear_call: Clearing call.
[4]ClearCallThread::ClearCallThread: Object
initialized.
[4]ClearCallThread::ClearCallThread: Unblock pipe - 7,
37
[2]WrapH323EndPoint::ClearCall: Request to clear call
[ip$localhost/32066]
[2]WrapH323Connection::OnSendReleaseComplete: Sending
RELEASE COMPLETE message [ip$localhost/32066]
[2]ClearCallThread::Main: Call with token
ip$localhost/32066 cleared.
[4]ClearCallThread::ClearCallThread: Object deleted.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[3]PAsteriskSoundChannel::Close: Closing os_handle 43
[3]PAsteriskSoundChannel::Close: Closing os_handle 45
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
I/Os: read=0, write=518
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
I/Os: write=0
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
I/Os: read=519, write=0
[3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
I/Os: write=0
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
deleted.
[2]WrapH323EndPoint::ClearCall: Request to clear call
[ip$localhost/32066]
[2]WrapH323EndPoint::OnConnectionCleared: Connection
[ip$localhost/32066] closed.
[2]WrapH323EndPoint::OnConnectionCleared: Call with
"GWCCS2K" completed
[4]WrapH323Connection::WrapH323Connection:
WrapH323Connection deleted.




	

	
		
___________________________________ 
Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB 
http://mail.yahoo.it

Daniel Varella de Oliveira

2005-Oct-31 06:21 UTC

head link

[Asterisk-Users] H323 one way audio using oh323

Mik,

 Your asterisk server is another machine of your GK ?  You can start verifying 
if the traffic between the machines (related to RTP packets) is ok.
 Do you have firewall ?
-- 

[ ]'s

Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
Rio de Janeiro - Brasil
www.tecnologiaip.com.br




On Monday 31 October 2005 08:05, mik sib wrote:> Hi all,
>
> through oh323 i can register to my gatekeeper and make
> and receive calls.
>
> My gatekeeper routes the incoming call as well as the
> outgoing.
>
> The problem is simply that i can't ear nothing from my
> SIP ipPhones. I can ear my voice from a normal
> telephone in my SIP phone but no viceversa.
>
> How can i debug this situation ? I've no errors in the
> log or at the asterisk startup.
> How to understand what's happening ?
> I've tryed different phones also.
> any idea ?
> thank you very much
> Mik
>
>
> Here's my oh323.conf
>  Configuration of OpenH323 channel driver
> ------------------------------------------
> Version: 0.7.3
> Listening on address: 10.0.0.253:1720
> Gatekeeper used: Nortel_H323_Gatekeeper@192.168.1.10
> (Registered)
> FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
> Supported formats in pref. order: ulaw<0>
> Jitter buffer limits (min/max): 20-100 ms
> TCP port range: 10000 - 20000
> UDP (RAS) port range: 10000 - 20000
> UDP (RTP) port range: 10000 - 20000
> IP Type-of-Service value: 0
> User input mode: rfc2833
> Max number of inbound H.323 calls: 100
> Max number of outbound H.323 calls: 100
> Max number of simultaneous H.323 calls: 100
> Max call rate (ingress direction): 1.00/30
> Default language: en
> Default music class: default
> Default context: voip-h323
>
> doing a call with the ip phone to the outside world
> through the gatekeeper
>
> [2]WrapperAPI::h323_make_call: Making call.
> [2]WrapH323EndPoint::MakeCall: Making call to
> 0258115040
> [4]WrapH323EndPoint::CreateConnection: Creating a
> H323Connection [32066]
> [2]WrapH323Connection::WrapH323Connection: Creation of
> WrapH323Connection based on user data.
> [2]WrapH323Connection::WrapH323Connection: Call is
> outgoing.
> [4]WrapH323Connection::WrapH323Connection:
> WrapH323Connection created.
> [3]WrapH323EndPoint::MakeCall: Call token is
> ip$localhost/32066
> [3]WrapH323EndPoint::MakeCall: Call reference is 32066
> [2]WrapH323Connection::OnSendSignalSetup: Sending
> SETUP message...
> [3]WrapH323Connection::OnSendSignalSetup: Setting
> display name 0432281316 Fabio Violino
> [3]WrapH323Connection::OnSendSignalSetup: Setting
> calling party number test419
> [2]WrapH323Connection::OnAlerting: Ringing phone for
> "0258115040" ...
> [3]WrapH323EndPoint::OpenAudioChannel: Direction =>
> RECODER, Buffer => 320
> [2]WrapH323EndPoint::OpenAudioChannel: Media format:
> FrameSize 8, FrameTime 8, TimeUnits 8
> [2]WrapH323EndPoint::OpenAudioChannel: Codec info:
> FrameRate 160
> [2]WrapH323EndPoint::OpenAudioChannel: Packet size:
> 160
> [2]WrapH323EndPoint::OpenAudioChannel: Frames per
> packet: 20
> [2]WrapH323EndPoint::OpenAudioChannel: LID Codec
> G.711-uLaw-64k
> [3]WrapH323EndPoint::OpenAudioChannel: The sound
> channel with the application is asterisk-oh323(fd=45)
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> initialized.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> initialized.
> [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
> Object initialized.
> [3]PAsteriskSoundChannel::Open: os_handle 45,
> mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
> 160
> [3]WrapH323EndPoint::OpenAudioChannel: Opened sound
> channel "Asterisk" for recording using 1x320 byte
> buffers.
> [3]WrapH323Connection::OnEstablished:
> WrapH323Connection [ip$localhost/32066] established
> (FastStartDisabled/noH245Tunneling)
> [3]WrapH323EndPoint::OnConnectionEstablished:
> Connection [ip$localhost/32066] established.
> [3]WrapH323EndPoint::GetConnectionInfo:
> [ip$localhost/32066] RTP Media:
> 10.0.0.253:10004-0.0.0.0:0
> [3]WrapH323EndPoint::OpenAudioChannel: Direction =>
> PLAYER, Buffer => 320
> [2]WrapH323EndPoint::OpenAudioChannel: Media format:
> FrameSize 8, FrameTime 8, TimeUnits 8
> [2]WrapH323EndPoint::OpenAudioChannel: Codec info:
> FrameRate 160
> [2]WrapH323EndPoint::OpenAudioChannel: Packet size:
> 160
> [2]WrapH323EndPoint::OpenAudioChannel: Frames per
> packet: 20
> [2]WrapH323EndPoint::OpenAudioChannel: LID Codec
> G.711-uLaw-64k
> [3]WrapH323EndPoint::OpenAudioChannel: The sound
> channel with the application is asterisk-oh323(fd=43)
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> initialized.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> initialized.
> [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
> Object initialized.
> [3]PAsteriskSoundChannel::Open: os_handle 43,
> mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
> 160
> [3]WrapH323EndPoint::OpenAudioChannel: Opened sound
> channel "Asterisk" for playing using 1x320 byte
> buffers.
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [5]PAsteriskSoundChannel::Read: Data read [320 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
> [3]WrapH323EndPoint::SetClearCallCause: Setting the
> Q.931 cause code of connection [ip$localhost/32066],
> at 16
> [2]WrapperAPI::h323_clear_call: Clearing call.
> [4]ClearCallThread::ClearCallThread: Object
> initialized.
> [4]ClearCallThread::ClearCallThread: Unblock pipe - 7,
> 37
> [2]WrapH323EndPoint::ClearCall: Request to clear call
> [ip$localhost/32066]
> [2]WrapH323Connection::OnSendReleaseComplete: Sending
> RELEASE COMPLETE message [ip$localhost/32066]
> [2]ClearCallThread::Main: Call with token
> ip$localhost/32066 cleared.
> [4]ClearCallThread::ClearCallThread: Object deleted.
> [5]PAsteriskSoundChannel::Write: Written [160 bytes]
> [3]PAsteriskSoundChannel::Close: Closing os_handle 43
> [3]PAsteriskSoundChannel::Close: Closing os_handle 45
> [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
> I/Os: read=0, write=518
> [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
> I/Os: write=0
> [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
> Object deleted.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> deleted.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> deleted.
> [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total
> I/Os: read=519, write=0
> [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Short
> I/Os: write=0
> [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
> Object deleted.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> deleted.
> [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
> deleted.
> [2]WrapH323EndPoint::ClearCall: Request to clear call
> [ip$localhost/32066]
> [2]WrapH323EndPoint::OnConnectionCleared: Connection
> [ip$localhost/32066] closed.
> [2]WrapH323EndPoint::OnConnectionCleared: Call with
> "GWCCS2K" completed
> [4]WrapH323Connection::WrapH323Connection:
> WrapH323Connection deleted.
>
>
>
>
>
>
>
>
> ___________________________________
> Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB
> http://mail.yahoo.it
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20051031/51c029dc/attachment.pgp

asterisk users - Oct 2005 - H323 one way audio using oh323

[Asterisk-Users] H323 one way audio using oh323

[Asterisk-Users] H323 one way audio using oh323