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