search for: h245tunneling

....! ;) 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 incominglimit=4 faststart=yes h245Tunneling=yes , my H323 softphone can't register. ("sent GRQ"..."gatekeeper not responding") My questions are : 1/ Can ooh323c work as gatekeeper (if yes, even if it is installed on the same box as asterisk)? 2/ if yes, Do you know tutorials for doing this? or Can anyone help...

Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. 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

I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912

...= { guid = 16 octets { 30 d4 bf de d3 8a d9 11 8e 4c 00 01 02 3f c2 76 0........L...?.v } } mediaWaitForConnect = FALSE canOverlapSend = FALSE multipleCalls = FALSE maintainConnection = FALSE } h245Tunneling = TRUE h245Control = 2 entries { [0]= 31 octets { 02 40 01 06 00 08 81 75 00 07 80 13 80 00 fa 00 .@.....u........ 01 00 00 01 00 00 01 00 00 0c c0 01 00 01 80 ............... } [1]= 7 octets { 01 00 32 80 3e 12 c6...

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

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone 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

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

...presentationAllowed = { 18:44:37:850 NULL 18:44:37:850 } 18:44:37:850 } 18:44:37:850 screeningIndicator = { 18:44:37:850 0 18:44:37:850 } 18:44:37:851 } 18:44:37:851 } 18:44:37:851 h245Tunneling = { 18:44:37:851 TRUE 18:44:37:851 } 18:44:37:851 } 18:44:37:851 UUIE decode successful 18:44:37:851 } 18:44:37:851 Queued H225 messages 1. (outgoing, ooh323c_o_10) 18:44:37:851 Sending H225 message (outgoing, ooh323c_o_10) 18:44:37:851 Sending Q931 message (outgoing, ooh3...

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

Send 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

...d9 88 84 88 36 4a e8 a3 4b .....%.....6J..K } callIdentifier = { guid = 16 octets { f2 97 9f 13 c9 25 11 d9 88 85 88 36 4a e8 a3 4b .....%.....6J..K } } fastConnectRefused = <<null>> } h245Tunneling = TRUE h245Control = 2 entries { [0]= 83 octets { 02 70 01 06 00 08 81 75 00 07 80 13 80 03 e8 00 .p.....u........ 01 00 00 01 00 00 01 00 00 0c c0 01 00 01 80 04 ................ 80 00 00 22 c0 17 80 00 01 22 80 17 80 00 02 83 ...&q...

...;Default - 1720 ;port=1720 ;The dotted IP address tvcti should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=192.168.22.224 ;Whether tvcti should use fast-start and tunneling for H323 connections. ;Default - yes ;faststart=no ;h245tunneling=no faststart=yes h245tunneling=yes ;H323-ID to be used for tvcti server ;Default - tvcti PBX ;h323id=ObjSystvcti ;e164=100 h323id=9 ;e164=100 ;e164=0,1,2,3,4,5,6,7,8,9,*,# ;CallerID to use for calls ;Default - Same as h323id callerid=tvcti ;Whether this tvcti server will use gatekeeper. ;Defa...

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

...s [internal] exten => 1000,1,Dial(SIP/1000) exten => 1000,2,HangUp() exten => _XXXX,1,Dial(H323/${EXTEN}@Avaya) exten => _XXXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya) exten => _XXXXXXXXXX,1,Dial(H323/${EXTEN}@Avaya) *Here is also the content of my ooh323.conf:* [general] faststart=yes h245tunneling=yes gatekeeper=DISABLE bindaddr=10.1.129.231 port=1720 callerID="ALT Asterisk PBX" progress_setup=8 progress_alert=8 disallow=all allow=all dtmfmode=inband faststart=yes context=internal [Avaya] type=friend context=internal host=10.1.129.247 port=1720 canreinvite=no disallow=all allow=al...

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

...c: -- User entered 'the pin number' Jan 19 10:00:43 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:8] GotoIf("DAHDI/2-1", "1?USER") in new stack My H323.conf file: [general] port=1720 bindaddr=ip address of asterisk gateway=no faststart=yes h245tunneling=yes h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=default disallow=all allow=ulaw dtmfmode=rfc2833 progress_setup = 8 progress_alert = 8 [denver] type=friend port=1720 ip=ip address of avaya context=from-internal disallow=all allow=ulaw rtptimeout=90 -...

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

...E [7177] logger.c: -- USER ENTERED 'THE PIN NUMBER' Jan 19 10:00:43 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:8] GotoIf("DAHDI/2-1", "1?USER") in new stack My H323.conf file: [general] port=1720 bindaddr=ip address of asterisk gateway=no faststart=yes h245tunneling=yes h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=default disallow=all allow=ulaw dtmfmode=rfc2833 progress_setup = 8 progress_alert = 8 [denver] type=friend port=1720 ip=ip address of avaya context=from-internal disallow=all allow=ulaw rtptimeout=90 --=_547786e...