Seems like it is the IAX jitterbuffer. Can anyone offer any insight as to why? If I turn jitterbuffer=no or disabled (comment it) then my one way audio after hold issue goes away.
On 11/8/06, Matt <[EMAIL PROTECTED]> wrote:
My iax.conf is: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=100 jittershrinkrate=1 notransfer=yes trunk=no [zoot] type=user secret=xxxxxxxx auth=plaintext host=zoot.xxxxxxxxxx.net notransfer=yes context=from-trunk On 11/8/06, Matt <[EMAIL PROTECTED]> wrote: > I (and some others) are having an issue with placing calls on hold. > > Our setup is as follows: > > IAX2 or SIP terminator/originator ---> asterisk box ---> SIP Phones > > I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have > the same issue. > > When I place a call on hold that has come in a PSTN channel (through a > PRI and a Digium card) everything is fine. When I place a call on > hold that has come in or gone out an IAX2 or SIP terminator or > originator, when I pick the call back up often there is one-way-audio > (I can hear the caller, but the caller can not hear me). > > I have attached a packet capture of the situation, and as you can see > at about packet #3803 the audio goes one way. Can anyone enlighten > me as to why this is happening, and why asterisk is no longer sending > audio back to the terminator? I would like to get this fixed, > obviously. > > (This file was a tcpdump, and can be opened in ethereal/wireshark) > > EDIT: Packet capture at following address: > http://hecate.chilitech.net/~matth/zootdump.cap >
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
