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)
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 >
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@lists.digium.com
[mailto:asterisk-users-bounces@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@leadtek.com.tw for seqno
102 (Non-critical Request)
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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). >
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)