Displaying 17 results from an estimated 17 matches for "jbimpl".
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
...d jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want to enable the jitter buffer for the end points having the
trouble.
Reading the docs, it seems that the jitter buffer is only used when the end
point is connected to an app like voicemail.
--
-Thermal
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2015 Mar 18
2
4 Port PRI
...idialplan=unknown
resetinterval=600
echocancel=yes
echotraining=yes
;echocancelwhenbridged=no
;rxgain=0
;txgain=0
callerid=asreceived
musiconhold=default
group=1
overlapdial=yes
signalling=pri_cpe
context=extensions
channel => 1-15,17-31
jbenable= yes
jbforce= yes
jbmaxsize= 120
jbimpl= fixed
jbresyncthreshold= 1000
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2009 Jul 22
1
grandstream and jitter buffer
...rs are
complaining they are jittery when I use "canreinvite=yes". The data
connection is an ADSL link dedicated for phone traffic. At any given
time, I have at most 2 calls in parallel.
I'm not a huge fan of asterisk being in media path doing buffering
because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and
sometimes my users complain that "are you on a sat phone?"
Any suggestions?
-
Kelvin Chan
2007 Dec 27
1
SIP Channel jitter buffer issue
...issues in its RTP Tx, which seems to have some amount of jitter
(mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and
max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf parameters jenable=yes, jbforce=yes,
jbmaxsize=200 and jbimpl=fixed. However on setting these parameters I am
unable to hear on the trunk side. From the jitter logs as given below, I can
see audio frames being dropped:
JB_PUT {now=1130}: Dropped frame with ts=21125 and len=20
JB_GET {now=1130}: now < next=2121
JB_GET {now=1142...
2015 Mar 18
1
4 Port PRI
...idialplan=unknown
resetinterval=600
echocancel=yes
echotraining=yes
;echocancelwhenbridged=no
;rxgain=0
;txgain=0
callerid=asreceived
musiconhold=default
group=1
overlapdial=yes
signalling=pri_cpe
context=extensions
channel => 1-15,17-31
jbenable= yes
jbforce= yes
jbmaxsize= 120
jbimpl= fixed
jbresyncthreshold= 1000
PRI or BRI? Which card are you using? Typically the installation script or
procedure lets you configure each span. You seem to have 4 spans for either
8 or 128 (EuroISDN) channels.
jg
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U...
2020 Feb 14
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
...Asterisk which is hosted in the cloud.
A typical sip.conf phone configuration on the remote server for the site is
[general]
session-timers=refuse
disallow=all
allow=g729:20
allow=ulaw
allow=alaw
fromuser=xxx
useragent=xxx
callcounter=yes
alwaysauthreject=yes
allowguest=no
jbnable=yes
jbforce=no
jbimpl=adaptive
jblog=no
jbmaxsize=200
jbresyncthreshold=1000
externaddr=xx.xx.xx.xx
localnet=xx.xx.xx.xx/255.255.255.0
[xxxx]
type=peer
user=xxxx
secret=xxxx
host=dynamic
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=xxxx
call-limit=1
limitonpeers=yes
callgroup=1
pickupgroup=1
d...
2015 Mar 18
0
4 Port PRI
...>
> ;txgain=0
>
> callerid=asreceived
>
> musiconhold=default
>
> group=1
>
> overlapdial=yes
>
> signalling=pri_cpe
>
> context=extensions
>
> channel => 1-15,17-31
>
> jbenable= yes
>
> jbforce= yes
>
> jbmaxsize= 120
>
> jbimpl= fixed
>
> jbresyncthreshold= 1000
>
PRI or BRI? Which card are you using? Typically the installation script or procedure lets you
configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels.
jg
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2008 Jan 25
0
Adaptive jitterbuffer problem
...jb.c: Failed to put first frame in the
jitterbuffer on channel ZAP".
This is my zapata.conf:
[trunkgroups]
[channels]
context=from-pbx
signalling=fxo_ks
usecallerid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
busydetect=yes
busycount=6
channel=>1
jbenable=yes
jbimpl=adaptive
Has anyone experienced such things? Any tips?
Thanks in advance, folks.
--
[]'s
Andr? de Abrantes
2008 Apr 30
0
Jitter buffer not used in SIP -> chan_local -> ZAP path even with /nj for local channels
...Working (jitter buffers created as expected):
ZAP -> SIP
SIP -> ZAP
Not working (no jitter buffers created):
SIP -> chan_local (with /nj) -> ZAP
SIP -> chan_local (with /j) -> ZAP
SIP -> chan_local (with no flags) -> ZAP
I have this in zapata.conf:
jbenable=yes
jbforce=no
jbimpl=fixed
jbmaxsize=300
Is there something I haven't tried that will make this work or will I have
to change my dialplan so it doesn't use local channels?
Thanks,
Mike
PS, here are some pages that I have used as sources of information:
No mention of /j for local channels
http://www.voip-i...
2009 Sep 08
0
Intermittent metallic voice SIP->ISDN ISDN<-SIP
...?
Thx
Pigi
The isdn is connected with an HFC-PCI card:
03:00.0 Network controller: Cologne Chip Designs GmbH ISDN network controller
[HFC-PCI] (rev 02)
this is my sip general part (jb enable to get the jitter buffer working):
jbenable = yes
jbforce = yes
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = adaptive
jblog = yes
This is the relevant part of the misdn-init.conf
card=1,hfcpci
te_ptp=1,2
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0
And this is the misdn.conf
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls...
2013 Jun 16
0
define extension to send calls to gatekeeper
...calls to
cisco gatekeeper.
this is my h323.conf file:
[general]
port=1720
binaddr=192.168.0.YY
context=from-trunk
faststart=yes
h245tunneling=yes
gatekeeper=192.168.0.XX //cisco address
progress_setup=8
progress_alert=8
dtmfmode=rfc2833
jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no
and this is extension that i defined for it in extensions.conf:
exten=>_2.,1,Dial(H323/2${EXTEN:1})
(if i define my extension like
(exten=>_2.,1,Dial(H323/to-cisco/2${EXTEN:1}), asterisk returns core dump
error.
but, i can't send my calls to cisco gatekeeper. do you have...
2010 Jan 15
1
jitterbuffer and PLC
...trunk,same result.
I don't know how to do. So please help me.
How to do to work correct.
Or Asterisk has not yet have jitter and PLC ,hasn't it?
In 'asterisk 1' ,
==================================================================
write on sip.conf => jbenable=yes ,and , jbimpl=adaptive
write on iax.conf => jitterbuffer=yes ,and, trunktimestamps=yes
write on codecs.conf => genericplc => true
and on extensions.conf
when use sip trunk =>
-------------------------------
exten => 3003,1,Dial(Local/3000 at extd/nj)
exten => 3000,1,Set(CALLERID(num)=...
2010 Nov 30
10
TCP port, VPN and resolving the cutting voice problem
Hi All;
Can I run the IAX on TCP port instead of UDP port?
If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)?
Same thing
2015 Jan 29
1
JITTERBUFFER function
> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to
call
> > the JITTERBUFFER function?
>
> You only need to use the JITTERBUFFER function.
>
> The jbenable option will enable a jitter buffer on every channel
> created for that peer (or, if global, for every peer in the system).
> Depending on the version of Asterisk, it will also place the
2020 Jun 23
2
Voice broken during calls (again...)
Am 23.06.2020 16:22, schrieb Marek Greško:
> It seems your problems lie in something other. Most probably it is not
> mtu problem. All my suspections are contradicted. If it is true you
> have inter vlan voice quality problems, it is definitely something
> different. Formerly I assumed you were trying only LTE vs LAN using
> internet.
I'm not sure what you mean with the last
2017 Feb 09
3
Disallow CALLS without registry
HI ALL
got small question
i use call-limit=1 on peers
but call limit is not working if user is not registered on PBX and
making calls
so the main question is -- how to Disallow CALLS without registering on PBX
--
Best regards
Antony
tel. +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
2008 Feb 08
1
(no subject)
...mestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with...