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...I did a *debug trc 10* on GnuGK and looked at the sessions... one from X-Lite through Asterisk... and one from OpenPhone... The one from OpenPhone works fine. And i found something on the beginning of the session... Here's the session from OpenPhone: 2004/09/21 14:52:35.219 2 RasSrv.cxx(2224) GK Read from 216.119.135.3:2367 2004/09/21 14:52:35.223 3 RasSrv.cxx(2237) GK admissionRequest { requestSeqNum = 40386 callType = pointToPoint <<null>> endpointIdentifier = 9 characters { 0032 0031 0032 0033 005f 0065 006e 0064 2123_end...

...3" timeStamp = 1139496380 token = { algorithmOID = 1.2.840.113549.2.5 paramS = { } hash = Hex: 06 d8 7b d1 ff 54 e3 bb c9 66 49 c2 4d cb 38 94 } } } willSupplyUUIEs = FALSE } 2006/02/09 15:47:02.165 1 RasSrv.cxx(343) RAS ARQ Received 2006/02/09 15:47:02.166 3 RasSrv.cxx(1948) GK ARQ will request bandwith of 1280 2006/02/09 15:47:02.181 2 RasSrv.cxx(388) ARJ|195.27.242.114:3775|100:dialedDigits|softpeer:h323_ID=123:dialedDigits|f alse|calledPartyNotRegistered; 2006/02...

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

...ugk Home=81.29.xxx.xxx;127.0.0.1 StatusTraceLevel=2 UseBroadcastListener=0 TimestampFormat=ISO8601 EndpointSignalPort=1720 EncryptAllPasswords=0 UseMulticastListener=0 StatusPort=7000 SignalCallId=1 [GkStatus::Auth] rule=password | explicit 81.29.xxx.xxx=allow Shutdown=allow gkadmin=eZqojUYXNUc= [RasSrv::PermanentEndpoints] 195.34.xxx.xxx:1720=gw1;00 81.29.xxx.xxx:1730=astgw;02 [RasSrv::GWPrefixes] gw1=00 astgw=02 [RasSrv::RRQFeatures] AcceptEndpointIdentifier=1 AcceptGatewayPrefixes=1 OverwriteEPOnSameAddress=1 IRQPollCount=0 [RasSrv::ARQFeatures] CallUnregisteredEndpoints=0 ArjReasonRouteCall...

...one receives ; "asterisk" as the caller-id [1004] type=h323 e164=011004 context=default [asterisk] type=h323 prefix=01 context=h323 /************************************************************************************ This is my entry in GnuGK [RasSrv::Neighbors] asterisk=68.90.233.134;1720;01; I've configured other GKs using this Neighbors section and it doesn't require password. Best regards, Carlos Maynard Jr. cmaynard@datatrans-inc.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists....

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083

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

...323 context=Office [7004] type=h323 context=Office ;[7005] ;type=h323 ;context=Office [gw1] type=h323 context=Office ----snip---- And here is my gatekeeper.ini file: [Gatekeeper::Main] Fourtytwo=42 Home=10.1.1.51 NetworkInterfaces=10.1.1.51/24 UseBroadcastListener=0 [GkStatus::Auth] rule=allow [RasSrv::GWPrefixes] gw1=2840 Does the above look correct. If I use Asterisk standalone and then connect ohPhone directly to Asterisk i can easily place outgoing calls, so the setup for outbound calls works I think. -- Mickey Binder

Hello folks. We are trying to interconnect an asterisk installation with a gnugk 2.0.5 installation to become able to use some H323 hardware that needs a gatekeeper (particulary an Ericsson WebSwitch 100). We have managed asterisk to dial H323 endpoints successfully (although calls are interrupted immediately after connection with "spawn extension exited non-zero"), but we could not

Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for