search for: jbresyncthreshold

...wn 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>

...ed. 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/pipermail/asterisk-users...

...nnection 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 -------------- An HTML at...

...wn 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 scrubbed... URL: <http://lists.digiu...

...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=xxx.xxx.xxx.xxx/255.255...

...; > 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 scrubbed... URL: <http://lists.digiu...

...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

...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 secret=1000 host=dynam...

...s. 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 ntdebugfile=/var/log/misdn-nt.l...

...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 ;qualify=300 directmedia...

...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 calls to cisco gatekeeper....

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

...; 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 in/broken timestamps, usualy sent from exotic devices ; an...