search for: jbmaxsize

Displaying 19 results from an estimated 19 matches for "jbmaxsize".

2009 Jul 22
1
grandstream and jitter buffer
...aw and my users 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
2015 Mar 18
2
4 Port PRI
...ype=euroisdn pridialplan=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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150318/098e52dd/attachment.html>
2011 Sep 14
1
Sip re-register / delay problem.
...and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/p...
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
...on a wireless connection that has very bad 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 -------------- next part -...
2014 May 27
0
dahdi-dahdi native bridging and audio level
...llo! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name display_receive=name ; display_receive= channel=>63-77,79-93 ;panasonic group=4 priindication = outofband rxwink=300 pridialplan=unknown prilocaldialplan=unkn...
2007 Dec 27
1
SIP Channel jitter buffer issue
...y client has some 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 J...
2015 Mar 18
1
4 Port PRI
...ype=euroisdn pridialplan=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 -------------- next part -------------- An HTML attachment was...
2020 Feb 14
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
...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=0.0.0.0/0.0.0.0 permit=x...
2015 Mar 18
0
4 Port PRI
...ed=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 -------------- next part -------------- An HT...
2007 Feb 07
0
Connection problem w/ Attended Transfer
...al(SIP/reception,10,t) ; at this point 'reception' [ie A] dials 100 exten => 100,1,Dial(SIP/mrblobby,10,t) ; the quality of the transferred call between mrblobby and exten => 100,2,Hangup ; voiptalk [ie B and C] is extremely poor sip.conf [general] jbenable = yes jbmaxsize = 1000 jbresyncthreshold = 1000 [reception] type=friend user=reception secret= callerid=Ben host=dynamic nat=no mailbox=100@default allow=all context=outgoing [mrblobby] type=friend user=mrblobby secret= callerid=Blobby host=dynamic nat=no mailbox=101@default allow=all context=outgoing
2008 Apr 30
0
Jitter buffer not used in SIP -> chan_local -> ZAP path even with /nj for local channels
...ter 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-info.org/wiki/ind...
2008 Nov 11
0
help with call with no sound via PSTN
...an_dahdi.conf trunkgroups] [channels] Group=1 context=incoming signalling=fxs_ks rxwink=300 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes jbenable=no jbmaxsize=200 jbresyncthreshold=1000 useincomingcalleridondahditransfer=yes ;callerid=asrecived rxgain=0.0 txgain=0.0 immediate=no busydetect=yes busycount=5 hidecallerid=no callgroup=1 pickupgroup=1 channel => 1-24 sip.conf [general] disallow=all allow=gsm allow=ulaw language=es [sets](!) type=friend...
2009 Sep 08
0
Intermittent metallic voice SIP->ISDN ISDN<-SIP
...right fix to this. Do you have any hint ? 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 ntdebug...
2012 Jan 13
1
Sporadic one way audio problem
...c of my server and the voice switch of my provider My sip.conf: [general] port=5060 bindaddr=0.0.0.0 language=de allowguest=no ;echocancel=yes ;echotraining=yes alwaysauthreject=yes disallow=all allow=alaw deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.X.X/29 permit=192.168.1.0/24 ;jbenable=yes ;jbforce=yes ;jbmaxsize=20 ;jbresyncthreshold=1000 tos=0x10 directmedia=no nat=no directrtpsetup=no [provider] type=peer host=XXX.XXX.X.X insecure=port,invite context=XXXXXXXXX dtmfmode=rfc2833 directmedia=no nat=no directrtpsetup=no ;qualify=300 [one-phone] [10] type=peer context=XXXXXXXXX secret=XXXXXX host=dynamic ;q...
2013 Jun 16
0
define extension to send calls to gatekeeper
...m and want to act as gateway and send 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 call...
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
2008 Feb 08
1
(no subject)
...; be used if the sending side can create jitter and jbforce is ; also set to yes. ; jbforce = no ; Forces the use of a ji tterbuffer on the receive side of a H323 ; channel. Defaults to "no". ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps...