Hi Pablo, I'm not really sure if you got an answer for this, but from what I can understand, you are trying to accomlish 2 issues currently not possible. The chan_h323 channel in asterisk doesn't support CLID forwarding correctly, so when a call goes out on H323 it will not have a CLID. You can forge a static CLID, with adding it to the end of h323.conf like this: [0123456789] type=h323 Now, regarding SIP to SIP, I need to understand a bit better, please elaborate as it sounds somewhat impossible. Nir S -----Original Message----- From: openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx [mailto:openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx] On Behalf Of pesb Sent: Thursday, April 15, 2004 5:28 PM To: asterisk-dev@xxxxxxxxxxxxxxxx Cc: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx; openh323gk-developer@xxxxxxxxxxxxxxxxxxxxx Subject: SIP to SIP calls authenticating through a GK(gnugk) Hi there, I am using asterisk as an H323 GW through the chan_h323 so that SIP terminals can interact with an H323 network managed by gnugk. When a SIP phone calls an H323 phone, everything works just fine. Except that the caller-id is not sent correctly. Now, what I need to do is to stablish a call from SIP to SIP authenticating through the H323 Gatekeeper. Is this possible? I have tried it, and it did not work. These are my config files: /*************************************************** h323.conf /*************************************************** [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=G723.1 allow=ulaw ; Allow codecs in order of preference allow=alaw gatekeeper = 192.168.0.103 context=h323 [1005] ; When this line and the context [1004] lines are set type=h323 ; the caller id 1004 is always sent. I don't know why. e164=011005 ; In case, this lines are not set, the GS phones receives context=default ; "Error" as the caller id, and the H323 phone receives ; "asterisk" as the caller-id [1004] type=h323 e164=011004 context=default [asterisk] type=h323 prefix=01 context=h323 /************************************************** extensions.conf (Just a few lines, the rest is the standar extensions.conf file) /************************************************** ;(...) [default] ;(...) exten => 021005,1,Dial(h323/011005) exten => 021004,1,Dial(h323/011004) ;exten => _03XXX,1,Dial(h323/${EXTEN:2}) exten => 301,1,Dial(h323/301) [h323] exten => 011005,1,Dial(SIP/1005) exten => 011004,1,Dial(SIP/1004) /************************************************** sip.conf /************************************************** [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls disallow=all ; Disallow all codecs allow=g729 allow=G723.1 allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret=123 host=dynamic canreinvite=no [1005] type=friend username=1005 secret=123 host=dynamic canreinvite=no /*****************************************************/ /*****************************************************/ Also, find attached an ethereal file with the call flow for a call from SIP to SIP through my Gatekeeper My scenario is the following: SIP Phones: Grandstream phones H323 Phone: Planet IP phone Gatekeeper: gnugk 1.0.7 Asterisk: version 0.9.0 As you can see, asterisk doesn't inform the calling partie that the called partie has picked up the phone, so no RTP conexion is stablished between asterisk and the calling partie. This happens always, without the matter of who is the calling partie(1004 or 1005) I am really in a hurry here. Please, somebody help me. Pablo Salinas ------------------------------------------------------- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149&alloc_id=8166&op=click _______________________________________________________ List: Openh323gk-users@xxxxxxxxxxxxxxxxxxxxx Archive: http://sourceforge.net/mailarchive/forum.php?forum_id=8549 Homepage: http://www.gnugk.org/
