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

[Matt]

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
>