search for: jbenable

> > 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...

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A calls B and I do Set(JITTERBUFFER(fixed)=default), my guess is that it will be atta...

Hello! 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 pr...

...ave a customer 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 ----------...

...annels] language=en switchtype=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>

Hi List, I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that jitterbuffer is only effective on the receiving channels. My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch office. Questions: 1. To enable jitter buffer on SIP channels it seems I have to enable and force it, right? 2. If I enable and force jitter buffer, Asterisk would always have to stay

...tsche Telekom uses since years just VoIP. No ISDN, PSTN, and so on... :( I'm connecting to the VoIP-Server of Deutsche Telekom via DSL (50Mbps down, 10Mbps up). The other party use VoIP, too, since they are in Germany (and Italy) and here there are just VoIP... Sigh! Now I disabled the jitter (jbenable = no), and I called my father in law. He sayd me, the quality is really better, but I hear sometimes little noises... Any other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

...5 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 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...

...wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [10000] type=friend secret=test host=dynamic nat=yes -------------------------- ----- extensions.conf: ----- [common] exten => 501,1,MeetMe(12,MI) exten => 501,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() --...

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

...5 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 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...

...annels] language=en switchtype=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 --------...

Hi, I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel => 1-23

...idecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no progzone=us tonezone=0 jbenable=yes ;Sangoma AFT-A200 [slot:4 bus:5 span:1] <wanpipe1> context=default group=0 echocancel=yes signalling=fxs_ks channel => 1 context=default group=0 echocancel=yes signalling=fxs_ks channel => 2 dahdi show version DAHDI Version: 2.1.0.4 Echo Canceller: MG2 dahdi show status Descrip...

...ses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no realm = myrealm.com udpbindaddr = 0.0.0.0 qualify = yes subscribecontext = default localnet=192.168.1.0/255.255.255.0 externhost=myhost.com externrefresh=30 dtmfmode = auto canreinvite = no jbenable = no sendrpid = yes trustrpid = no disallow=all allow=ulaw allow=alaw register => myuser:mypass at vpsserver [vpsserver] type=friend secret=myuser defaultuser=mypass host=vpsserver.domain.com context=inbound canreinvite=no insecure=port,invite And this is the server's pjsip.conf [transpor...

I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like

...ibecontext = device-hints allowexternaldomains = 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 rel...

Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten => _X.,1,Answer() exten => _X.,n,Wait(n) exten => _X.,n,...do stuff... exten => _X.,n,Hangup() exten => fax,1,Goto(fax-rx,receive,1) [fax-rx] exten => receive,1,... exten =>

...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....

...1225808102,2,Dial(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=out...