Re: [Asterisk-Users] Video phone settings???
Ronald_Wiplinger wrote: I tried this and it echos the picture back on Xten but not on the hard phones, ... What codecs are you using on the hardphones? Does the voice echo back on the hardphones? Do they have a button to start video? What bandwith rate are they using? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
Re: [Asterisk-Users] Video phone settings???
Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. Giorgio. Ronald_Wiplinger wrote: 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:
Re: [Asterisk-Users] Video phone settings???
Giorgio Incantalupo wrote: Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. I have already set that. Without that NO video at all at any try. bye Ronald Giorgio. Ronald_Wiplinger wrote: 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
RE: [Asterisk-Users] Video phone settings???
Hi, -Original Message- disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Have you tried permutations of this ? I have had working setups with everything except h263p. My experience with leadtek phones is they tend to crash when they are talking to any phone model that is not exactly the same (i.e. bugs in decoding perhaps ?) Florian ___ 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
Re: [Asterisk-Users] Video phone settings???
Hi, It seems that you are using different audio codec (Unknown RTP codec 96 received) Try to use standard audio code. Sometimes telephone use codec with bad rtp code inside. I use alw or ulaw for my test. Marino On 7/11/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. I have already set that. Without that NO video at all at any try. bye Ronald Giorgio. Ronald_Wiplinger wrote: 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,
RE: [Asterisk-Users] Video phone settings???
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
Re: [Asterisk-Users] Video phone settings???
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
Re: [Asterisk-Users] Video phone settings???
apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? bye Ronald Wiplinger 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
Re: [Asterisk-Users] Video phone settings???
Ronald Wiplinger wrote: apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? Just add a line to your extensions.conf: exten = 600,1,Echo() And that should do it. Also try the hardphones with different resolutions/bandwidths (CIF/QCIF). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
Re: [Asterisk-Users] Video phone settings???
Matt Riddell wrote: Ronald Wiplinger wrote: apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? Just add a line to your extensions.conf: exten = 600,1,Echo() I tried this and it echos the picture back on Xten but not on the hard phones, ... bye Ronald And that should do it. Also try the hardphones with different resolutions/bandwidths (CIF/QCIF). ___ 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
