[Asterisk-bsd] Re: One Way Audio

[Konstantin Prokazoff]

Welcome everyone!

    I have returned.
    Does anyone have working on wct4xxp porting to BSD?
    I take one for mine call-center, but don't wanna to migrate to pinguin.
;)

BR,
    Oryx
tel. +38 044 244 11 81, fax. +38 044 234 04 55

----- Original Message -----
From: "Jeff Rizzo" <riz+asterisk at boogers.sf.ca.us>
To: "Asterisk on BSD discussion" <asterisk-bsd at lists.digium.com>
Sent: Saturday, October 16, 2004 6:05 PM
Subject: Re: [Asterisk-bsd] Re: One Way Audio


> Tom Ivar Helbekkmo wrote:
>
> >Jay Adelson <jay at adelson.org> writes:
> >
> >
> >
> >>If the remote side calls my phone, particularly with reinvite on, I
> >>can hear the remote side fine and the call works normally.
> >>
> >>
> >
> >Well, since the call runs through an Asterisk that has one foot on the
> >outside, with real addresses, and one on the inside, with the phones
> >on, and NAT in between, it's important that canreinvite=no.  Otherwise,
> >the phone end points will try to speak RTP directly, and while your
> >phone is then able to address packets to the real IP address on the
> >other end, that end can't reach your phone through NAT.
> >
> >
> >
>
> (I've been working with Jay on this problem for a few days now - a bit
> of the blind leading the blind :-)
>
> One of the interesting things about this setup is that the _only_
> situation where audio is actually two-way is where canreinvite=yes is
> set on both his extension (1000) and the connection to sipphone.com, and
> he calls _out_.  When I do a tcpdump on his inside interface, the thing
> that I notice is that in the half-duplex audio situation, packets from
> the phone to asterisk are destined for one of the interfaces on the
> asterisk box, and packets from asterisk to the phone are destined for
> the other interface, which makes me wonder if the phone itself (a
> polycom 500) is ignoring them.
> Audio _does_ work both ways when he uses a softphone (SJphone), even
> though the same situation of packets from the phone going to one
> interface, and packets to the phone come from the other.
>
> >You may also have a codec selection problem.  You should probably
> >clean out the passwords from a copy of your sip.conf, and post it
> >here.  At the same time, check which codecs your phones support, and
> >whether they can be set up with preferences for codec choice.
> >
> >
> >
> We did play with codecs a little, but as I said we're still kind of
> stumbling around in the dark here.   :)
>
> +j
>
> >-tih
> >
> >
>
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