Displaying 20 results from an estimated 10000 matches similar to: "[HELP] Regarding stripping of fmtp parameters for Video."
2007 Jul 12
0
No subject
supported by Asterisk for Video.
I also find that video_caps branch has a fix for this problem, please can
someone share more information
about this and where i can find it ?
I do not want those fmtp lines to be stripped. Suggestions to change the
Asterisk config files, to achieve this are also welcome.
Thank you.
Best regards,
Simith
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2010 Dec 06
1
Asterisk 1.6.2.10 video call
Hello list,
I'm trying to set up a video call from my Ekiga client to a Grandstream
GXV3140 IP-phone. The call succeeds but there is no video.
I have in sip.conf :
videosupport=yes
disallow=all
allow=alaw
allow=g726
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
The Grandstream peer has codecs (sip.conf) :
gsm;alaw;g729;h261;h263;h263p;h264
The Ekiga peer has codecs
2010 Aug 09
0
[SIP/H.264] Codec negotiation problem ?
Hi,
I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : "every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated"
What I observe :
- a call made from a SIP Phone registred in Asterisk to Tandberg works
(voice and video bidirectionnal)
- a call
2007 Sep 20
0
Video doesn't work for outgoing call?
I've tried to put a call file to /var/spool/asterisk/outgoing/
to make an outgoing video call, but not succeeded.
I could hear the audio, but no video.
The asterisk version is 1.4.10, with videosupport=yes
The client is eyebeam 1.5.7, with h263 support.
Here are some debug messages.
It shows the client and asterisk negotiated the video capabilities
without problem. However, the 'show
2011 Dec 02
1
Where to download sample video file for Asterisk 1.8x playback?
Hello,
I have been trying to playback a video file via Playback() in Asterisk
1.8.7.1 but the file format seems to fail.
[2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File
/etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format
[2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to
open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4
2009 Oct 23
3
SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
the Record function parameters should be?
In sip.conf I have:
disallow=all
2014 Mar 31
1
Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways
We are experiencing an issue with our Cisco 9971 and 8945 phones where H264
video calls are connecting at 176x144 resolution instead of 640x480. Soft
clients can connect at higher resolutions and the 9971 can even receive
video at a higher resolution (although it still sends 176x144).
I contacted one of the developers and he suggested the passthrough of SDP
attributes is not working correctly.
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with "make samples",
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same trouble.
I have several Uniden UIP200 phones and a single Grandstream BudgetTone
2018 Feb 21
0
AST-2018-003: Crash with an invalid SDP fmtp attribute
Asterisk Project Security Advisory - AST-2018-003
Product Asterisk
Summary Crash with an invalid SDP fmtp attribute
Nature of Advisory Remote crash
Susceptibility Remote Authenticated Sessions
Severity Minor
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
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I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2014 Aug 06
1
From and To headers contain same account in INVITEs
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I have 660 at testers.com as
a websocket client and 700 at testers.com (caller) using a Zoiper client (db
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root at elx4 ~]#
2009 Sep 02
1
outbound calls not ringing still
i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me.
INVITE sip:+185993133333 at 216.82.224.202
2007 Apr 26
1
Can asterisk record the duration of users putting on hold?
Hi,
Recently we got a new feature request from our customer, they want a
report to list the duration that agents putting customer on hold, they
want to base on this to measure the agents performance. I cannot find
any events in cdr, message logs, or manager interface, only when I
enable sip debug, then I can see the ReInvite Event in the cli , some
thing like the attached logs, is there any
2019 Apr 17
2
IPv6 transport results in ICE with only IPv6 candidates
Hi,
I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport:
[transport-tcp-ipv6]
type=transport
protocol=tcp
bind=[2001:1234:5678:abcd::2]:5060
I also have an IPv4 version of that:
[transport-tcp-ipv4]
type=transport
protocol=tcp
bind=10.75.22.8:5060
I've then configured an endpoint to use it:
[outgoing]
type = endpoint
context = default
dtmf_mode = none
disallow = all
2014 Dec 11
0
PJSIP configuration question
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP ---
<--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP
2014 Dec 14
0
PJSIP configuration question
Trying this again after my first away from work in a couple weeks.
Running Asterisk 13.0.0
IP authentication with Vitelity
I can Originate with sip, but not pjsip.
Here is the sip settings and trace.
Action: Originate
ActionID: S8
Channel: SIP/8005555555 at outbound.vitelity.net
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable:
2014 Oct 27
0
Codec setting using fmtp maxaveragebitrate and OPUS_SET_BITRATE
Hi Folks, thanks for the great work, not sure if this is the right list
for this type of quesiton.
We are looking to use only Opus as "one codec for all", with VoIP-out
obviously we want to tune it.
I am planning to use fmtp in SDP to control server/client Opus settings.
Something like :
- *maxplaybackrate*: a hint about the maximum output sampling rate that
the receiver is
2006 Mar 03
0
a=fmtp:18 annexb=no
Hello
Looking the SIP debug we see a change in the SETUP
message from the Asterisk 1.0.x version to the 1.2.4.
In the 1.2.4 we notice this line:
a=fmtp:18 annexb=no
This line cause problems in our plattform (We think
our SIP -> h323 gateway can't parse this line)
Why this line its present in 1.2.4 version?
Have anybody some clue?
Regards
JS.
2005 Mar 15
0
dial to h.323
hello
i want to rout my calls to h.323. i have registered my
asterisk with GnuGatekeeper. but it is not routing my
call to h.323 channel. he is saying Internal channel
initialization failed. Bad binary?
can any one check my settings what is problem here
thanks in advance
kamran
exten=>_321XXXX,1,Dial(OH323/${EXTEN}@192.168.0.153:1719,30,r)