[Asterisk-Users] Video phone settings???

[Ronald_Wiplinger]

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)