[Asterisk-Users] Video phone settings???

map mapunt at gmail.com
Mon Jul 11 02:41:48 MST 2005


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 <ronald_wiplinger at leadtek.com.tw> 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, 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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> 
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