search for: h245tunnelling

Hello, I have asterisk 1.4 with ooh323c addons installed. (As I am a newbie in voip world...my question might be idiot...! ;) Please forgive me!) I succeed to make H323 call when ooh323c is configured as gateway (gatekeeper=DISABLE in ooh323.conf). When I put gatekeeper= ip_address, and add an account as follow : [aaa] type=friend username=aaa password=xxxx host=dynamic context=test

...urces. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver ------------------------------------------ Version: 0.7.1 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: g729<0> Jitter buffer limits (min/max): 20-500 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: 3 Max number of inbound H.323 calls:...

...need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912 ID - HMA0200.10szxn-xxxx e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 gatekeeper=AVS@210.21.118.XXX gatekeeperTTL=600 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec...

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

...penphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 amaFlags=default accountCode=H323 [...

...perServer::WrapGatekeeperServer: Creating new gatekeeper. Ouch ... error while writing audio data: : Broken pipe Segmentation fault ------------ oh323.conf ---------------------------- [general] listenAddress=myip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=5 libTraceFile=/var/log/asterisk/oh323.log gatekeeper=mygnugk ;gatekeeperPassword=secret gatekeeperTTL=600 userInputMode=TONE...

...ons and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=30000 udpStart=20001 udpEnd=30000 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 lib...

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

...ring with gatekeeper "200.69.192.252". Sep 29 00:07:32 ERROR[-151090528]: chan_h323.c:1987 load_module: Gatekeeper registration failed. my h323.conf is: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=200.69.192.252 gatekeeperPassword = secret name=xxxxxxxxxx secret=xxxxxxxxxx AllowGKRouted...

...d these questions to Asterisk-Users mailing list. h323.conf ################################################## ; ; Configuration file of OpenH323 channel driver ; [general] listenAddress=W.X.Y.Z ; local ip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=yes jitterMin=20 jitterMax=100 ipTos=none outboundMax=100 inboundMax=100 simultaneousMax=100 wrapLibTraceLevel=0 libTraceLevel=0 libTraceFile=stdout gatekeeper=A.B.C.D ; GnuGK gatekeeperTTL=600 ; Set the mode for sending user-input (DTMF) ; Valid values for this option are:...

>Please I have combed the Archive to no avail on this problem protocol >control problem in oh323. >I'm receiving calls from CISCO AS5300 -> Asterisk -> Zap Channel. The >calls clears the remote location but drops on my own end. Please what >could be >wrong. I have included the oh323.conf and log files. I have tried >various configuration and I thought I should

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems Thanks a million Seshu, it worked like a champ. Thanks >From: "Kanuri, Seshu (Company IT)" <Seshu.Ka...

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

...o be used by H.323 ; tcpStart=10000 tcpEnd=20000 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; "rtp.conf" ; udpStart=10000 udpEnd=20000 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...10000). ; jitterMin=20 jitterMa...

...,1,Dial(SIP/519,20,Tt) exten => 102,2,Hangup exten => 102,102,Hangup [local-access] include => extensions ------------- h323.conf ----------- [general] listenAddress=xx.xx.xx.xx,xx listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DISCOVER gatekeeperTTL=600 userInputMode=RFC2833 amaFlags=default accountCode=H323 context=v...

Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:7] Read("DAHDI/2-1", "PIN|enter-conf-pin-number||||") in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- <DAHDI/2-1> Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- User entered

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:7] Read("DAHDI/2-1", "PIN|enter-conf-pin-number||||") in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- USER ENTERED 'THE PIN NUMBER' Jan 19 10:00:43 VERBOSE [7177] logger.c: --