Displaying 8 results from an estimated 8 matches for "jblog".
Did you mean:
blog
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
...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
-------------- next part --------------
An HTML attachment was scrubbed...
URL...
2020 Feb 14
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
...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
deny=0.0.0.0/0.0...
2010 Nov 03
1
inbound call issue...
...lowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = yes
checkmwi = 10
compactheaders = no
defaultexpiry = 120
dumphistory = no
externip = 216.26.109.22
g726nonstandard = no
jbenable = yes
jbforce = no
jblog = no
localnet = internal subnet
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = no
sendrpid = no...
2009 Sep 08
0
Intermittent metallic voice SIP->ISDN ISDN<-SIP
...sdn 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=no
bridging=no
l...
2002 Jul 18
1
NT user name resolution to UNIX user name doesn't work the same
My departmental file server is on RHL6.2(i386) (Samba 2.0.6) and participates in the company's NT domain. If an NT user wants to see shares on this UNIX(Linux) box I have only to create a UNIX account for them where the UNIX username matches the NT username. I have a Solaris 8 box (Samba 2.2.2), I cannot figure out how to make it act the same way. I have looked at winbindd but that does noot
2013 Jun 16
0
define extension to send calls to gatekeeper
...co 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 any suggest...
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
2008 Feb 08
1
(no subject)
...; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
&nb sp; ; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;...