Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem.
Regards. On 7/11/05, Storm D. J. Petersen <[EMAIL PROTECTED]> wrote: > I found the problem was with eyeBeam when I had more than one video codec > enabled. Try on eyebeam to only have h263p enabled. > > Does the video appear in the Echo test? > > S. > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ronald_Wiplinger > Sent: Monday, July 11, 2005 12:41 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Video phone settings??? > > I have three video phones here for testing: > > Extension 6003 is Eyebeam > Extension 6004 is a hard phone (model 8770) > Extension 6005 is a hard phone (model 8882) > > Can anybody have a look at my settings and the output I get from all > kinds of dialings, please. > > The sip settings for all phones is (user / password different): > > [6003] > type=friend > username=6003 > secret=pwd > qualify=200 > nat=yes > host=dynamic > canreinvite=yes > context=from-sip > callerid=Ronald Wiplinger <6003> > dtmfmode=rfc2833 > disallow=all > allow=ulaw > allow=alaw > allow=h261 > allow=h263 > allow=h263p > > > > > > > Tests on 7/11/2005 > > Eybeam to 8770 > > both screens are black!!! > > > e*CLI> > -- Executing Dial("SIP/6003-94ec", "SIP/6004|60|trm") in new stack > -- Called 6004 > -- Started music on hold, class 'default', on SIP/6003-94ec > -- SIP/6004-4b4d is ringing > -- SIP/6004-4b4d answered SIP/6003-94ec > -- Stopped music on hold on SIP/6003-94ec > -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d > == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' > > > > -------------- > > Eybeam to 8882 > > both screens are black!!! > > > *CLI> > -- Executing Dial("SIP/6003-8a2e", "SIP/6005|60|trm") in new stack > -- Called 6005 > -- Started music on hold, class 'default', on SIP/6003-8a2e > -- SIP/6005-fa6a is ringing > -- SIP/6005-fa6a answered SIP/6003-8a2e > -- Stopped music on hold on SIP/6003-8a2e > -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a > == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' > > > > -------------- > > 8770 to 8882 > > both screens are black!!! > > > *CLI> > -- Executing Dial("SIP/6004-5e88", "SIP/6005|60|trm") in new stack > -- Called 6005 > -- Started music on hold, class 'default', on SIP/6004-5e88 > -- SIP/6005-5180 is ringing > -- SIP/6005-5180 answered SIP/6004-5e88 > -- Stopped music on hold on SIP/6004-5e88 > -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' > > > > -------------- > > 8770 to Eyebeam > > 8770 gets picture, Eybeam no picture > > > -- Executing Dial("SIP/6004-5e88", "SIP/6005|60|trm") in new stack > -- Called 6005 > -- Started music on hold, class 'default', on SIP/6004-5e88 > -- SIP/6005-5180 is ringing > -- SIP/6005-5180 answered SIP/6004-5e88 > -- Stopped music on hold on SIP/6004-5e88 > -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec > 96 received > == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' > -- Executing Dial("SIP/6004-2cff", "SIP/6003|60|trm") in new stack > -- Called 6003 > -- Started music on hold, class 'default', on SIP/6004-2cff > -- SIP/6003-322c is ringing > -- SIP/6003-322c answered SIP/6004-2cff > -- Stopped music on hold on SIP/6004-2cff > -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c > == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' > > -------------- > > 8882 to Eyebeam > > both screens are black!!! > > > -- Executing Dial("SIP/6005-3361", "SIP/6003|60|trm") in new stack > -- Called 6003 > -- Started music on hold, class 'default', on SIP/6005-3361 > -- SIP/6003-9ed0 is ringing > -- SIP/6003-9ed0 answered SIP/6005-3361 > -- Stopped music on hold on SIP/6005-3361 > -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 > > > -------------- > > 8882 to 8770 > > 8882 gets a picture > > > -- Executing Dial("SIP/6005-abd3", "SIP/6004|60|trm") in new stack > -- Called 6004 > -- Started music on hold, class 'default', on SIP/6005-abd3 > -- SIP/6004-6381 is ringing > -- SIP/6004-6381 answered SIP/6005-abd3 > -- Stopped music on hold on SIP/6005-abd3 > -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 > == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' > Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum > retries exceeded on call [EMAIL PROTECTED] for seqno > 102 (Non-critical Request) > > > _______________________________________________ > 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