search for: admissionrequest

...penPhone... 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 0070 p } destinationInfo = 1 entries { [0]=dialedDigits "83521...

...lls in the reverse direction. I have tried many different configs on the GK, but always come up with the same error. It appears to me that asterisk successfully registers with GK as I can see the aliases and the e.164 numbers, but when the h323 softphone tries to call my IAX softphone, I get this: admissionRequest { requestSeqNum = 2 callType = pointToPoint <<null>> endpointIdentifier = 9 characters { 0033 0032 0039 0037 005f 0065 006e 0064 3297_end 0070 p } destinationInfo = 1 entries { [0]=dialedDigits "100"...

...eeper=x.x.x.x (gk ip) allowgkrouted=yes allow=all [xx] (my login) type=h323 e164=111 (my assigned phone number) secret=1111 (my password) context=incoming I'm getting this error when trying to make a call (h.323 trace 1) 2:37.884 Transactor:814d8d0 h323trans.cxx(678) Trans admissionRequest rejected: callerNotRegistered 2:56.764 Transactor:814d8d0 h323trans.cxx(678) Trans registrationRequest rejected: fullRegistrationRequired 4:06.543 Transactor:814d8d0 h323trans.cxx(678) Trans registrationRequest rejected: fullRegistrationRequired It seems that...

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

...c7bb0 RFC2833 Handler created 1:17.531 ThreadID=0x445c7bb0 H323 Added capability: G.729{hw} <1> 1:17.531 ThreadID=0x445c7bb0 H323 Created new connection: ip$localhost/18577 1:17.531 H225 Caller:9e2d880 H225 Started call thread 1:17.535 H225 Caller:9e2d880 Trans Making request: admissionRequest 1:17.535 H225 Caller:9e2d880 Trans Sending PDU: admissionRequest 37497 1:17.535 H225 Caller:9e2d880 Trans Waiting on response to seqnum=37497 for 3.0 seconds 1:17.655 Transactor:9df23b8 H225RAS Receiving PDU: admissionConfirm 37497 1:17.656 H225 Caller:9e2d880 H323 Bandwidt...

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