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