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: [EMAIL PROTECTED] > (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 > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
pgpP96O4TqObQ.pgp
Description: PGP signature
_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
