[Asterisk-Users] H323 still no rtp traffic
Hi all, i'm still experiencing a one way call only between a ipPhone and an analog one through a oh323 channel between my asterisk and a Nortel GK. Doing some sniffing and some debug with ethereal and tcpump i can say (i hope, as newby to say the right thing) that i can't see any rtp traffic between the asterisk and the nortel. In the analog phone (in the outside telecom world) i can't ear nothing said in the ipPhone. Viceversa in the ipPhone (Mitel one) i can ear the voice comming from the outside world. In my sip.conf [419] callerid=0432281316 TEST test 419 type=friend username=419 secret=password host=dynamic nat=yes canreinvite=no reinvite=no disallow=all allow=ulaw allow=gsm ;allow=alaw dtmfmode=rfc2833 context=out callgroup=1 pickupgroup=1 There's no rtp traffic from the phone or from the asterisk to the GK. The GK stays on the intranet even if it has a internet looking ip. ipPhone 10.24.3.40 asterisk 10.24.2.253 GK 80.74.178.196 Issuing on asterisk rtp debug [2]WrapH323EndPoint::AnswerCall: Request to answer call ip$80.74.178.196:34404/1169 Got RTP packet from 10.24.3.40:20012 (type 0, seq 14, ts -1120604096, len 160) [2]WrapH323EndPoint::AnswerCall: Call answered [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 15, ts -1120603936, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 16, ts -1120603776, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [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=42) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 42, 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$80.74.178.196:34404/1169] established (FastStartDisabled/H245Tunneling) [3]WrapH323EndPoint::OnConnectionEstablished: Connection [ip$80.74.178.196:34404/1169] established. [3]WrapH323EndPoint::GetConnectionInfo: [ip$80.74.178.196:34404/1169] RTP Media: 10.24.2.253:21002-0.0.0.0:0 Got RTP packet from 10.24.3.40:20012 (type 0, seq 17, ts -1120603616, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 18, ts -1120603456, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 19, ts -1120603296, len 160) [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=40) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 40, 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] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26203, ts 160, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [5]PAsteriskSoundChannel::Read: Data read [320 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 20, ts -1120603136, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26204, ts 320, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 21, ts -1120602976, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet
[Asterisk-Users] one way audio on oh323 channel, there's no rtp traffic
Hi all, i'm experiencing a one way call only between a ipPhone and an analog one through a oh323 channel between my asterisk and a Nortel GK. Doing some sniffing and some debug with ethereal and tcpump i can say (i hope, as newby to say the right thing) that i can't see any rtp traffic between the asterisk and the nortel. In the analog phone (in the outside telecom world) i can't ear nothing said in the ipPhone. Viceversa in the ipPhone (Mitel one) i can ear the voice comming from the outside world. In my sip.conf [419] callerid=0432281316 TEST test 419 type=friend username=419 secret=password host=dynamic nat=yes canreinvite=no reinvite=no disallow=all allow=ulaw allow=gsm ;allow=alaw dtmfmode=rfc2833 context=out callgroup=1 pickupgroup=1 There's no rtp traffic from the phone or from the asterisk to the GK. The GK stays on the intranet even if it has a internet looking ip. ipPhone 10.24.3.40 asterisk 10.24.2.253 GK 80.74.178.196 Issuing on asterisk rtp debug [2]WrapH323EndPoint::AnswerCall: Request to answer call ip$80.74.178.196:34404/1169 Got RTP packet from 10.24.3.40:20012 (type 0, seq 14, ts -1120604096, len 160) [2]WrapH323EndPoint::AnswerCall: Call answered [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 15, ts -1120603936, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 16, ts -1120603776, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [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=42) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 42, 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$80.74.178.196:34404/1169] established (FastStartDisabled/H245Tunneling) [3]WrapH323EndPoint::OnConnectionEstablished: Connection [ip$80.74.178.196:34404/1169] established. [3]WrapH323EndPoint::GetConnectionInfo: [ip$80.74.178.196:34404/1169] RTP Media: 10.24.2.253:21002-0.0.0.0:0 Got RTP packet from 10.24.3.40:20012 (type 0, seq 17, ts -1120603616, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 18, ts -1120603456, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 19, ts -1120603296, len 160) [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=40) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 40, 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] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26203, ts 160, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [5]PAsteriskSoundChannel::Read: Data read [320 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 20, ts -1120603136, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26204, ts 320, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 21, ts -1120602976, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to
[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: [EMAIL PROTECTED] (Registered) FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported formats in pref. order: ulaw0 Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 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:
[Asterisk-Users] h323 no audio from the sip phone to the outside world.
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 during a call 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: ulaw0 Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 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]
[Asterisk-Users] H323 REGISTRATION PROBLEM: Gatekeeper '[EMAIL PROTECTED] ' found but failed to register
Hi all
First of all excuse me if i make such a big post, hope
also to write in the right place.
I need to connect my linux/asterisk (10.0.0.252) box
to a Nortel PBX (192.168.1.10) with h323
I'd like to allow some phones to register via sip to
asterisk and
with these to the Nortel PBX wich gives me the
connections to the outside world (phone)
after downloading and compiling the latest asterisk
source from cvs
OpenH323 v1.15.6, PWlib v1.8.7 (Mimas version from
Voxgratia)
and oh323-0.7.3 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz
starting asterisk i get
[4]WrapProcess::Main: Starting...
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time
libraries OpenH323 v1.15.6, PWlib v1.8.7
[2]WrapperAPI::h323_end_point_create: Endpoint
created.
[3]WrapperAPI::h323_set_options: Setting endpoint
options.
[3]WrapperAPI::h323_set_ports: Setting endpoint port
ranges.
[2]WrapperAPI::h323_removeall_capabilities: Removing
all capabilities.
[3]WrapH323EndPoint::RemoveAllCapabilities: Removing
all capabilities of local endpoint.
[5]WrapH323EndPoint::SetFrames: Setting 20
[5]WrapH323EndPoint::GetFrames: Returning 20
[2]WrapperAPI::h323_set_capability: Inserted
capability G.711-ALaw-64k{hw}
[3]WrapperAPI::h323_set_senduimode: User-input mode
set.
[2]WrapperAPI::h323_set_gk: Configuring gatekeeper.
[3]WrapH323EndPoint::SetGatekeeperTimeToLive:
Gatekeeper registration TTL set at 600 sec
[4]GKRegThread::GKRegThread: Object initialized.
[4]GKRegThread::GKRegThread: Unblock pipe - 20, 21
[3]WrapperAPI::h323_callback_register: Callback
functions installed.
[2]GKRegThread::Main: GK: name [192.168.1.10], zone []
[2]GKRegThread::Main: Failed to register with GK name
[192.168.1.10], zone []
[4]WrapperAPI::h323_get_gk: Checking gatekeeper.
-- Gatekeeper '[EMAIL PROTECTED]'
found but failed to register
RAS Failed registration of with
Nortel_H323_Gatekeeper
i'm wondering three things.
FIRST QUESTION
Am'i right in the idea? is asterisk capable the
realize what i need ?
SECOND QUESTION
the guy working in the telco said me that i can see on
the Nortel pbx the connection attempt
but from 127.0.0.1. By reading the oh323.log i can see
that during the RAS phase my asterisk
send the loopback address
in the following log i can see
rasAddress = 1 entries {
[0]=ipAddress {
ip = 4 octets {
7f 00 00 01
}
port = 10002
}
}
0:00.145 GKRegThread:0816ac30 TCP Appending H.225
transport ip$10.0.0.253:1720 using associated
transport Transport[remote=ip$192.168.1.10:1719
if=ip$127.0.0.1:10001]
THIRD QUESTION
why in the string
RAS Failed registration of with
Nortel_H323_Gatekeeper
after the word of there's only a blank space?
thank you very much for your patience and for your
precious help (i hope)
in the oh323.log
0:00.007 asterisk-oh323 H323Created
endpoint.
0:00.029 H323 Cleaner H323Started
cleaner thread
0:00.029 asterisk-oh323 H323Started
listener Listener[ip$10.0.0.253:1720]
0:00.030 asterisk-oh323 H323Added
capability: G.711-ALaw-64k{hw} 1
0:00.030 asterisk-oh323 H323Added
capability: UserInput/hookflash 2
0:00.030 asterisk-oh323 H323Added
capability: UserInput/basicString 3
0:00.030 asterisk-oh323 H323Added
capability: UserInput/dtmf 4
0:00.030 asterisk-oh323 H323Added
capability: UserInput/RFC2833 5
0:00.054H323 Listener:816a698 H323Awaiting TCP
connections on port 1720
0:00.054H323 Listener:816a698 TCP Waiting on
socket accept on ip$10.0.0.253:1720
0:00.054 GKRegThread:0816ac30 H323UDP Binding to
interface: :::10001
0:00.056 GKRegThread:0816ac30 RAS Authenticator
H235AnnexD_Procedure1no-pwd not active during GRQ
SetCapability negotiation
0:00.056 GKRegThread:0816ac30 RAS Authenticator
CATno-pwd not active during GRQ SetCapability
negotiation
0:00.056 GKRegThread:0816ac30 RAS Authenticator
MD5no-pwd not active during GRQ SetCapability
negotiation
0:00.056 GKRegThread:0816ac30 H225Started
gatekeeper discovery of ip$192.168.1.10
0:00.056 GKRegThread:0816ac30 RAS Searching
interfaces:
127.0.0.1
[00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:01]
00-00-00-00-00-00 (lo)
10.0.0.253
[fe:80:00:00:00:00:00:00:02:01:02:ff:fe:12:02:92]
00-01-02-12-02-92 (eth0)
0:00.056 GKRegThread:0816ac30 RAS Gatekeeper
discovery on interface: 10.0.0.253:10002
0:00.057GkMonitor:816cae0 RAS Background
thread started
0:00.086 GKRegThread:0816ac30 Trans Sending PDU:
gatekeeperRequest {
requestSeqNum = 65022
protocolIdentifier = 0.0.8.2250.0.4
rasAddress = ipAddress {
ip = 4 octets {
0a 18 02 fd
}
port = 10002
