[asterisk-users] Re: One-Way-Audio After placing call on hold

[Matt]

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 <mhoppes at gmail.com> 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 <mhoppes at gmail.com> 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
> >
>