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...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

...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>

...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/...

...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...

...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=unk...

...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...

...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...

...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=...

...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 H...

...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

...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/in...

...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...

...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 ntdebu...

...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 ;...

...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 cal...

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

> > 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

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

...; 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 jump...