Sebastian Mangelkramer
2005-Sep-11 03:11 UTC
[Asterisk-Users] OpenH323-Channel Q.931-Problems with Gatekeeper
Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk with the Gatekeeper (Siemens Surpass) of an big german Carrier we noticed some strange problems we couldn`t solve until right now. The registration with the gatekeeper is successful. But every from and to our PBX will be cleared/rejected by an Q.931 cause. Our system-layout looks like: Debian GNU/Linux 3.1 aka "sarge" with Kernel 2.6.12, i386 Asterisk 1.0.9 (stable) Pwlib 1.16 OpenH323 1.13.5 Chan_oh323 0.6.6 Perhaps you know some problems with Asterisk and the H.323-Channel. We tried to compile and test nearly every version of openh323 and chan_oh323, but it wasn`t successful. Best regards from Germany, Sebastian. Nearby we will post our configs and logs: 1.) chan_oh323.conf ----------------------------- [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 language=de ; erweitertes logging aktivieren (debugging) wrapLibTraceLevel=9 libTraceLevel=9 libTraceFile=/var/log/asterisk/oh323.log ; gatekeeper des carrier gatekeeper=XXX.XXX.XXX.XXX gatekeeperTTL=600 userInputMode=TONE ; detailierte cdr erstellen amaFlags=billing accountCode=0123456789 ; eingehende calls an diesen context senden context=carrier-in [register] context=carrier-in alias=0123456789 [codecs] codec=G711A frames=20 2.) Status of OpenH323 channel driver --------------------------------------- *CLI> oh323 show conf Version: 0.6.6 Listening on address: 0.0.0.0:1720 Gatekeeper used: RRS@XXX.XXX.XXX.XXX (Registered) FastStart/H245Tunnelling/H245inSetup: ON/OFF/OFF Supported formats in pref. order: alaw<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: 2 Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Default language: Default music class: Default context: h323-in 3.) Verbose debugging of OpenH323 channel driver while calling from carrier ----------------------------------------------------------------------------- ------------------------------ *CLI> [4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [1797] [4]WrapH323Connection::WrapH323Connection: WrapH323Connection created. [2]WrapH323Connection::OnReceivedSignalSetup: Received SETUP message... [2]WrapH323Connection::OnAnswerCall: User ----- (016097XXXXXX) [IP of Carrier-GK] is calling us... [3]WrapH323Connection::OnAnswerCall: Call ID: 02cb6411-b5a7-178c-2499-0800062a0cf1 [3]WrapH323Connection::OnAnswerCall: Conference ID: 02cb6411-b5a7-178c-2499-0800062a0cf1 [3]WrapH323Connection::OnAnswerCall: Call reference: 1797 [3]WrapH323Connection::OnAnswerCall: Call token: ip$IP of Carrier-GK:36031/1797 [3]WrapH323Connection::OnAnswerCall: Call source alias: ----- (016097XXXXXXX) [IP of Carrier-GK](35) [3]WrapH323Connection::OnAnswerCall: Call dest alias: 0123456789 0123456789 E164:123456789 ip$10.0.0.20:1720(64) [3]WrapH323Connection::OnAnswerCall: Call source e164: 016097XXXXXX(12) [3]WrapH323Connection::OnAnswerCall: Call dest e164: 0123456789(13) [3]WrapH323Connection::OnAnswerCall: Call RDNIS: (0) [3]WrapH323Connection::OnAnswerCall: Remote Party number: 016097XXXXXXXX [3]WrapH323Connection::OnAnswerCall: Remote Party name: ----- (016097XXXXXXXX) [IP of Carrier-GK] [3]WrapH323Connection::OnAnswerCall: Remote Party address: 016097XXXXXX@ip$IP of Carrier-GK:36031 [3]WrapH323Connection::OnAnswerCall: Remote Application: Surpass Siemens 4/130(21) Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797' detected. -- Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797' detected. Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797', channel 'OH323/-----@IP of Carrier-GK-d66d'. -- Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797', channel 'OH323/-----@IP of Carrier-GK-d66d'. [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-ALaw-64k Setting channel 'OH323/-----@IP of Carrier-GK-d66d' (ip$IP of Carrier-GK:36031/1797) native format to alaw! [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=44) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 44, mediaFormat 8, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel "Asterisk" for recording using 1x320 byte buffers. [3]WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=42) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 42, mediaFormat 8, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel "Asterisk" for playing using 1x320 byte buffers. -- Executing Dial("OH323/-----@IP of Carrier-GK-d66d", "SIP/0123456789|30") in new stack Sep 9 20:47:18 NOTICE[5183]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing VoiceMail("OH323/-----@IP of Carrier-GK-d66d", "u5007263") in new stack [2]WrapperAPI::h323_answer_call: Answering call. [2]WrapH323EndPoint::AnswerCall: Request to answer call ip$IP of Carrier-GK:36031/1797 [2]WrapH323EndPoint::AnswerCall: Call answered [ip$IP of Carrier-GK:36031/1797] Channel OH323/-----@IP of Carrier-GK-d66d answered. -- Playing 'vm-theperson' (language 'en') [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [3]WrapH323EndPoint::GetConnectionInfo: [ip$IP of Carrier-GK:36031/1797] RTP Media: 10.0.0.20:10000-0.0.0.0:0 [5]PAsteriskSoundChannel::Write: Written [160 bytes] Channel OH323/-----@IP of Carrier-GK-d66d (call 'ip$IP of Carrier-GK:36031/1797') RX byte count is 160. [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [5]PAsteriskSoundChannel::Write: Written [160 bytes] [5]PAsteriskSoundChannel::Read: Data read [320 bytes] [2]WrapH323Connection::OnReceivedReleaseComplete: Received RELEASE COMPLETE message [ip$IP of Carrier-GK:36031/1797] [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$IP of Carrier-GK:36031/1797] [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$IP of Carrier-GK:36031/1797] [5]PAsteriskSoundChannel::Write: Written [160 bytes] [3]PAsteriskSoundChannel::Close: Closing os_handle 42 [3]PAsteriskSoundChannel::Close: Closing os_handle 44 [3]PAsteriskSoundChannel::PAsteriskSoundChannel: Total I/Os: read=0, write=3 [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=4, 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::OnConnectionCleared: Connection [ip$IP of Carrier-GK:36031/1797] closed. Call 'ip$IP of Carrier-GK:36031/1797' cleared. -- H.323 call 'ip$IP of Carrier-GK:36031/1797' cleared, reason 24 (Call ended with Q.931 cause [21 - Call rejected]) [2]WrapH323EndPoint::OnConnectionCleared: Call with "----- (016097XXXXXX) [IP of Carrier-GK]" completed [4]WrapH323Connection::WrapH323Connection: WrapH323Connection deleted. Sep 9 20:47:18 WARNING[5183]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (carrier-in, 0123456789, 2) exited non-zero on 'OH323/-----@IP of Carrier-GK-d66d' -- Hungup 'OH323/-----@IP of Carrier-GK-d66d' Call 'ip$IP of Carrier-GK:36031/1797' without owner has already been cleared (2). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050911/11789ec7/attachment.htm