quickly this looks like a incompatible codec.. or unrecognized.. show codecs on CLI>
show show 262144 (1 << 18) (0x40000) video h261 (H.261 Video) 524288 (1 << 19) (0x80000) video h263 (H.263 Video) 1048576 (1 << 20) (0x100000) video h263p (H.263+ Video) does it ? On 8/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > 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 > _______________________________________________ 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