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 to 10.24.3.40:20012 (type 0, seq 26205, ts 480, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 22, ts -1120602816, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26206, ts 640, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 23, ts -1120602656, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26207, ts 800, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 24, ts -1120602496, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26208, ts 960, len 160) snip [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 45, ts -1120599136, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26229, ts 4320, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 46, ts -1120598976, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26230, ts 4480, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 47, ts -1120598816, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26231, ts 4640, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26232, ts 4800, len 160) [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26233, ts 4960, len 160) [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26234, ts 5120, len 160) [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [5]PAsteriskSoundChannel::Write: Written [160 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [5]PAsteriskSoundChannel::Write: Written [160 bytes] snip [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] [3]WrapH323EndPoint::SetClearCallCause: Setting the Q.931 cause code of connection [ip$80.74.178.196:34404/1169], at 16 [2]WrapperAPI::h323_clear_call: Clearing call. [4]ClearCallThread::ClearCallThread: Object initialized. [4]ClearCallThread::ClearCallThread: Unblock pipe - 45, 46 [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnSendReleaseComplete: Sending RELEASE COMPLETE message [ip$80.74.178.196:34404/1169] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [2]ClearCallThread::Main: Call with token ip$80.74.178.196:34404/1169 cleared. [4]ClearCallThread::ClearCallThread: Object deleted. [5]PAsteriskSoundChannel::Write: Written [160 bytes] [3]PAsteriskSoundChannel::Close: Closing os_handle 40 [3]PAsteriskSoundChannel::Close: Closing os_handle 42 [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total I/Os: read=0, write=62 [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=64, 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$80.74.178.196:34404/1169] [2]WrapH323EndPoint::OnConnectionCleared: Connection [ip$80.74.178.196:34404/1169] closed. [2]WrapH323EndPoint::OnConnectionCleared: Call with "E164:0432707350 [80.74.178.196]" completed [4]WrapH323Connection::WrapH323Connection: WrapH323Connection deleted. I can't see rtp traffic from asterisk to the gk ! Today i've changed the rtp.conf and oh323.conf file to tell my asterisk to use rtp ports from 21000 to 30000. issuing oh323 show conf Configuration of OpenH323 channel driver ------------------------------------------ Version: 0.7.3 Listening on address: 10.24.2.253:1720 Gatekeeper used: Nortel_H323_Gatekeeper@80.74.178.196 (Registered) FastStart/H245Tunnelling/H245inSetup: OFF/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: 21000 - 30000 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 and during the test i can see issuing tcpdump -i eth0 -n host 80.74.178.196 snip 17:23:14.441390 IP 80.74.178.196.34396 > 10.24.2.253.1720: P 852:913(61) ack 794 win 10887 17:23:14.442622 IP 10.24.2.253.1720 > 80.74.178.196.34396: F 794:794(0) ack 913 win 6432 17:23:14.469007 IP 80.74.178.196.34396 > 10.24.2.253.1720: . ack 795 win 0 17:23:14.469416 IP 80.74.178.196.34396 > 10.24.2.253.1720: F 913:913(0) ack 795 win 0 17:23:14.469435 IP 10.24.2.253.1720 > 80.74.178.196.34396: . ack 914 win 6432 17:23:14.470624 IP 10.24.2.253.10002 > 80.74.178.196.1719: UDP, length: 175 17:23:14.501858 IP 80.74.178.196.1719 > 10.24.2.253.10002: UDP, length: 3 17:23:38.970892 IP 10.24.2.253.10002 > 80.74.178.196.1719: UDP, length: 372 17:23:39.011764 IP 80.74.178.196.1719 > 10.24.2.253.10002: UDP, length: 137 17:24:04.015093 IP 10.24.2.253.10002 > 80.74.178.196.1719: UDP, length: 372 17:24:04.058344 IP 80.74.178.196.1719 > 10.24.2.253.10002: UDP, length: 137 17:24:29.061334 IP 10.24.2.253.10002 > 80.74.178.196.1719: UDP, length: 372 17:24:29.106223 IP 80.74.178.196.1719 > 10.24.2.253.10002: UDP, length: 137 no traffic from ports in the range from 21000 to 30000 is made !!! ROUTING AND FIREWALL The telco also handles my wan and they say me that there're no firewall/routers that drop the traffic. I don't really trust very much in them ! CODECS The phone, asterisk and the gk all use ulaw (G.711 u-law) Other sip calls between sip phones and sip and analog (locallly configured zaptel interfaces) are normally working on the same asterisk Any idea ? Mik ___________________________________ Yahoo! Messenger: chiamate gratuite in tutto il mondo http://it.messenger.yahoo.com