Displaying 19 results from an estimated 19 matches for "dvi4".
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2010 Mar 17
9
[Bug 27136] New: blank screen with G98 [Quadro NVS 420] (NV98) dual GPU, 4-head
http://bugs.freedesktop.org/show_bug.cgi?id=27136
Summary: blank screen with G98 [Quadro NVS 420] (NV98) dual GPU,
4-head
Product: xorg
Version: 7.5
Platform: x86-64 (AMD64)
OS/Version: Linux (All)
Status: NEW
Severity: normal
Priority: medium
Component: Driver/nouveau
2011 Mar 02
0
Asterisk 1.6 and windows RTC
....9.130.
s=session.
c=IN IP4 172.31.9.130.
b=CT:1000.
t=0 0.
m=audio 4632 RTP/AVP 97 111 112 6 0 8 4 5 3 101.
k=base64:ftJemQZ2pTDV5gzzqxG6ps5Ol5qiOt2qbP9L9Or5JQg.
a=rtpmap:97 red/8000.
a=rtpmap:111 SIREN/16000.
a=fmtp:111 bitrate=16000.
a=rtpmap:112 G7221/16000.
a=fmtp:112 bitrate=24000.
a=rtpmap:6 DVI4/16000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:5 DVI4/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=encryption:optional.
a=direction:active.
OK from asterisk 1.6 PBX:
v=0.
o=PBX 1705093286 1705093286 IN IP4 172.31.9.251.
s=PBX....
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
...th: 448
v=0
o=root 5903 5903 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes
17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/...
2006 Jan 17
2
IAX/SIP and openser problem. IAX bug?
.....m=audio 14640 RTP/AVP 0 8 4 111 18
3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap
:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:111
G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 G
SM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110
speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 teleph
one-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
U 2006/01/16 12:21:10.969161 10.2.11.35:5060 -> 10.2.11.35:5062
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
10.2.11.35:5062;branch=z9hG4bK31f811a3;rport=5062..From: "Zyxel"
<sip:Zyxel<sip:20
4@1...
2004 Jun 03
4
miserable time with Cisco ATA186
..., BYE, REFER
Content-Type: application/sdp
Content-Length: 461
v=0
o=root 284 284 IN IP4 munged
s=session
c=IN IP4 munged
t=0 0
m=audio 14466 RTP/AVP 0 8 4 3 2 5 10 7 18 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
This Retransmits several times and then the call is scheduled for
destruction. The "CANCEL" sip...
2008 Sep 28
1
G.722 between Eyebeam and a Polycom IP650
Hi All,
So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes as "HD"
mode and the audio quality is certainly very good.
However, things are not so easy with Eyebeam and the
2003 Aug 01
1
Asterisk SIP bug with Net2Phone
...application/sdp
Content-Length: 384
v=0
o=root 21604 21604 IN IP4 192.0.0.0
s=session
c=IN IP4 192.0.0.0
t=0 0
m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:14 MPA/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
(no NAT) to 66.33.146.12:5060
-- Called 1800XXXXXXX@net2phone
Sip read:
SIP/2.0 407 Unauthorized
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<s...
2005 Feb 17
1
Voicepulse Open Access & Asterisk Problems
...nt-Length: 373
v=0erisk*CLI>
o=root 24964 24964 IN IP4 66.234.228.159
s=session
c=IN IP4 66.234.228.159
t=0 0
m=audio 10602 RTP/AVP 0 8 3 110 97 2 5 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
11 headers, 16 lines
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 5
Found RTP audio fo...
2004 Jan 14
1
Codec matching weirdness
...one-event/8000
a=fmtp:101 0-15
Sent to remote server by *
v=0
o=root 4205 4205 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 16798 RTP/AVP 4 3 0 8 2 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 192.246.69.223:5060
Received from remote server
v=0
o=root 9755 9756 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m...
2005 Jul 10
3
Incoming calls from BudgetPhone.nl
...Jul 2005 16:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 345
?
v=0
o=root 26318 26318 IN IP4 212.203.28.2
s=session
c=IN IP4 81.23.228.139
t=0 0
m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
?
--- (15 headers 15 lines)---
Using INVITE request as basis request -
3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl
Sending to 81.23.228.15...
2005 Jul 10
0
(no subject)
...37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 345
v=0
o=root 26318 26318 IN IP4 212.203.28.2
s=session
c=IN IP4 81.23.228.139
t=0 0
m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 15 lines)---
Using INVITE request as basis request -
3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl
Sending to 81...
2005 Mar 17
2
Snom190 intercom
...ication/sdp
Content-Length: 467
v=0
o=root 8153 8153 IN IP4 203.30.X.Y
s=session
c=IN IP4 203.30.X.Y
t=0 0
m=audio 10850 RTP/AVP 8 0 3 97 4 2 5 10 7 18 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
2007 Feb 21
3
SIP 406 error - cause?
...ported: replaces
Content-Type: application/sdp
Content-Length: 467
v=0
o=root 5921 5921 IN IP4 99.99.26.93
s=session
c=IN IP4 99.99.26.93
t=0 0
m=audio 16738 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 1123459913134831424@voicemaster.domain.net
<--- SIP read from 10...
2019 Apr 17
2
IPv6 transport results in ICE with only IPv6 candidates
...andidate:Hdccea0f2 2 UDP 2130706430 2001:123:ab:123::2 14385 typ host
a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 14385 typ host
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32...
2008 Mar 10
0
Audiocodes MP124-FXS replying BUSY when line is not.
...orted: replaces
Content-Type: application/sdp
Content-Length: 471
v=0
o=root 31887 31887 IN IP4 aa.bb.cc.dd
s=session
c=IN IP4 aa.bb.cc.dd
t=0 0
m=audio 47732 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- Called 90166
[Mar 10 11:13:14] DE...
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2014 Dec 11
0
PJSIP configuration question
....2.0
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 13752 RTP/AVP 10 4 3 0 8 111 5 7 18 110 117 97 112 9 118 102 115 116 119 107 96 108 109 113 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221...
2011 Feb 10
2
Unable to make outgoing calls with Internode
...ngth: 738
v=0
o=root 51098296 51098296 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0
m=audio 19850 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/ -- Called 0871271201 at sip-out
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
From: "<my ata cid>" <sip:<phone number...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain