Displaying 20 results from an estimated 700 matches similar to: "Jitterbuffer issues"
2006 Feb 20
9
Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using G729 with Asterisk 1.07. With the new ability to do packet
loss correction with ILBC, I felt I'd give it a try. The new PLC does
not work with G729. I don't use Speex because my softphone does not
support it.
This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569
(IAX2). I've never really stressed the bandwidth. Typically, only
10-20 concurrent calls.
2006 Jan 22
2
Disposition codes in CDR
Is there any way to have more specific disposition codes in the CDR?
Currently there are only 3 values: ANSWER, NO ANSWER, BUSY.
In this way, when i call a cell phone that is switched off i get "NO
ANSWER", while i would like to be able to log that the call is not
answered because "The customer you have dialed is unavailable at the
moment".
The same for "non
2009 Oct 16
0
Origin of "Exceptionally long voice queue length queuing to IAX2/blahblah" messages
Hello,
I'm using asterisk for a quite long period, i integrated a lot of stuff to make it behave like any carrier class system, so users can:
manage Call forward on busy
manage Call forward on no answer
manage unconditionnal call forward
call back missed calls
and a lot of such services that can be both configured by menus and dtmf or by web interfaces (PHP+SQL), in fact i recreated
2013 Jan 25
1
Frames with invalid timing info
I'm now getting these errors:
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP
even
2008 Jan 25
0
Adaptive jitterbuffer problem
Hello there,
I have set a simple environment to test some functionalities of
asterisk's new jitterbuffer.
The environment is composed of a sip softphone registering in asterisk
1.4 and calling a pstn phone connected to asterisk through a fxs
board.
Using the fixed buffer implementation the call quality is improved
when injecting a artificial jitter in my local network. However, when
changing
2013 Jan 24
5
"clicking" sound with alaw codec
I'm trying to interface Asterisk with an Alcatel PABX and trying to find
a code that works well. It says it doesn't support ulaw, though it
doesn't reject it. It supports G.729, and that works fine, but we'd prefer
not to use compression.
When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
2012 Jan 18
1
Compile error 1.8.8.1
Hi,
While compiling 1.8.8.1, I met the following error:
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o
2011 Mar 07
1
[1.8.3] Error compiling Asterisk: __sync_fetch_and_add
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting this
error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o
2007 Jan 08
3
jitterbuffer on sip.conf
In iax.conf there is option jitterbuffer
how about sip protocol ? Are jitterbuffer can configure in sip.conf ?
Thanks, for your share
2005 Oct 07
1
Distorted VM with iax2 with ilbc and jitterbuffer - bug?
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax config'ed as:
trunk=yes
allow=ilbc
jitterbuffer=yes
Recorded VM messages are very distorted.
Changing only
2006 Nov 16
0
jitterbuffer in pure voip (sip/iax) - what is best practice
I know, that jitterbuffer should be set at receiving side and on
outgoing call leg,
ie. if sipphone calls to asterisk and outgoing to zap chanel, I should
set jitterbuffer on zap channel (to dejjitter audio stream from sipphone)
but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)?
I have following setup (homeworkers using sip phone connected to home
asterisk via SIP and
2015 Jan 29
0
JITTERBUFFER function
On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson
<torbjorn.abrahamsson at gmail.com> wrote:
> Hello!
>
>
>
> I am going to use the JITTERBUFFER function in a SIP (and local channels)
> only setup, but have some questions of how to use it:
>
>
>
> 1. Do I need to activate jbenable in sip.conf? Or is it enough to call
> the JITTERBUFFER function?
2006 Jan 25
1
jitterbuffer causes no sound?
Hi guys,
I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at
the third days I activated setting jitterbuffer=yes and suddenly there
is no voice when the call is picked up. It's really weird as if asterisk
stops sending rtp packet. I've checked asterisk log and found nothing
suspicious. Just weird :S.
I tried it in 3 asterisk server and all of them are having
2006 Feb 27
2
jitterbuffer and DTMF conflict?
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too: http://bugs.digium.com/view.php?id=6011
Can anyone suggest a workaround (other than jitterbuffer=off)?
- Mike
2010 Jan 15
1
jitterbuffer and PLC
Hi, I have a question about jitterbuffer and PLC.
I use Asterisk 1.6.2.0 and 1.6.0.20 or older.
I use uLaw.
My system map:
=============================================================================
[ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]
=============================================================================
I use two asterisk server.
2006 May 25
2
jitterbuffer causes flaky IAX2 incoming connections?
I've been having problems with incoming IAX2 calls - some work, but a
large fraction are answered with "dead air" or disconnects from my IAX
provider.
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
provider is using.
A snippet of the a failed incoming call IAX2 debug is attached
2009 Sep 01
4
jitterbuffer for chan_sip on asterisk 1.2
Hello,
2015 Feb 18
1
SIP Jitterbuffer
Hello people
What are the cons, if any, of enabling a jitterbuffer?
We are currently using version 1.8
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY
2014 Jan 22
3
running LXC hello world example
Hello,
I am having difficulty getting any sort of LXC container running. I am
trying to use the following tutorial to run the hello world example:
https://www.berrange.com/posts/2011/09/27/getting-started-with-lxc-using-libvirt/
Here are the results of my running the tutorial:
[root@terraria ~]# virsh list
Id Name State
2007 Nov 05
0
JitterBuffer in SVN
Thorvald Natvig a ?crit :
> I see you're changing the jitter buffer around quite a bit. Could you
> let us know when it's ready for general testing? (At the moment it
> doesn't handle missing packets at all)
While I'm not completely done yet, I thought the current version was
working. Can you tell me what happens exactly (without output if
possible) so I can fix it? Also,