Thank you for your answer. I didn't register on the domain of the Eyebeam software, actually I don't understand how to do that! I bouught 5 eyebeam activation keys and I am trying with the first 2 of them
On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263" codec, no other. If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the two video phone speak without any problem (but without any video) If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the first video phone call the second, the second answer and immediately the call ends. If Ilook at /var/log/asterisk/full, I see: ........ Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi completed, returning 0 Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0", "SIP/552|25|tr") in new stack Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL) Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0 Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0 Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x80000 formats Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552 Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0 Aug 17 08:37:06 VERBOSE[14731]: -- Called 552 Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102 Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop: <sip:[EMAIL PROTECTED]:5060> Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0 Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2) to SIP/552-ff46(524288) Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make SIP/551-eac0 compatible with SIP/552-ff46 Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse counter It seems the problem documented in bug http://bugs.digium.com/bug_view_page.php?bug_id=0003709 but actually it is not exactly the same. moreover: is there any way to put the patch described in http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *) in asterisk 1.0.9 and not asterisk CVS HEAD ? Any help will be greatly appreciated. Andrea "Carlos Alperin" <[EMAIL PROTECTED] om.net> To Sent by: "'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion'" [EMAIL PROTECTED] <asterisk-users@lists.digium.com> m.com cc Subject 16/08/2005 20.48 RE: [Asterisk-Users] problems with eyebeam - video phone Please respond to Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED] ists.digium.com> Hi, I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I only use H.263 and SIP. (G.729) Now, the more important question is if you register on the domain on the Eyebeam software. I found that this was the full secret about this. Let me know your configuration on the Eyebeam side. Regards, Carlos Alperin -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 11:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problems with eyebeam - video phone I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ; allow=h263p ; H.263p is the enhanced video codec context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf And I left only H.263 basic in codec's configuration in Video Phone. No chance to get the communication in H.263 protocol. I saw that to use H.263+ protocol I need Asterisk CVS. I am not using asterisk CVS I am using asterisk 1.0.9 (last stable version a couple of week ago..) Is there any chance to make asterisk 1.0.9 to support SIP video calls in eyeBeam ? Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users