[Asterisk-Users] Video phone settings???

Giorgio Incantalupo
Mon, 11 Jul 2005 00:52:38 -0000

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 2002fb00-4d0b478-13c4@leadtek.com.tw 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
>