[Asterisk-Users] Video phone settings???

[Ronald Wiplinger]

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 <stormp at telus.net> 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: asterisk-users-bounces at lists.digium.com
>>[mailto:asterisk-users-bounces at lists.digium.com] 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 2002fb00-4d0b478-13c4 at leadtek.com.tw for seqno
>>102 (Non-critical Request)
>>
>>
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>>
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>


-- 
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com    +886 (0) 939--77-55-16  or FWD 511208
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