[Serusers] RTP Proxy Help - One way Audio

Vitaly Nikolaev vitaly at cifnet.com
Wed Feb 18 01:16:05 CET 2004


And of course put:

log(1, "blablabla"); and all if, else, etc in config....



On Tue, 2004-02-17 at 12:09, Darren Nay wrote:
> I removed it and I still have the same problem.  My onreply_route looks like
> this now.
> 
> onreply_route[1] {
>     if (status =~ "(183)|2[0-9][0-9]") {
>         fix_nated_contact();
>         force_rtp_proxy();
>     } else if (nat_uac_test("1")) {
>         fix_nated_contact();
>     };
> }
> 
> I also tried this..
> 
> onreply_route[1] {
>         fix_nated_contact();
>         force_rtp_proxy();
> }
> 
> I must admit that I'm new to all this SIP routing.  :(
> 
> One more thing of note is that we are getting the following errors in the
> syslog.
> 
> Feb 17 12:53:15 lvl3 /usr/local/sbin/ser[19711]: ERROR: send_rtpp_command:
> can't read reply from a RTP proxy
> Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: extract_body:
> message body has lenght zero
> Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: force_rtp_proxy:
> can't extract body from the message
> Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: on_reply processing
> failed
> 
> I just noticed this on my last call attempt.
> 
> Any more suggestions.  I've tried all kinds of combinations .. I'm stumped..
> 
> Thanks!
> Darren
> 
> 
> ----- Original Message ----- 
> From: "Maxim Sobolev" <sobomax at portaone.com>
> To: "Darren Nay" <dnay at libertyisp.com>
> Cc: <serusers at lists.iptel.org>
> Sent: Tuesday, February 17, 2004 2:41 PM
> Subject: Re: [Serusers] RTP Proxy Help - One way Audio
> 
> 
> > As I said in my previoud message, most likely that isflagset(6) test in
> > onreply_route is causing problems. Remove it completely and try again.
> >
> > -Maxim
> >
> > Darren Nay wrote:
> >
> > > Hey Guys,
> > >
> > > I am having issues with One way Audio for outgoing phone calls from my
> > > SIP phones.  It works fine for Incoming, but outgoing audio is not
> > > working.  Also, Outgoing works fine if I put my SIP phone on an Internet
> > > IP address, but if it's NAT'd then I get the 1 way audio on outgoing
> calls.
> > >
> > > I am using the default nathelper module config, but I have hacked it a
> > > bit and maybe my changes are causing the problem?  I had to use
> > > t_relay_to_tcp for our PSTN gateway and so I had to change the routing
> > > around a little.
> > >
> > > Prior to today I was using SerMediaProxy (AG Projects), but I switched
> > > to PortaOnes rtpproxy just to make sure that there wasn't a version
> > > incompatibility problem.
> > >
> > > I have also attached my current ser.cfg file.
> > >
> > > Anyone have any suggestions?  Or should I change the routing in my
> > > ser.cfg file?  or would that make a difference?
> > >
> > > Any help would be greatly appreciated.  Thanks!
> > >
> > > Darren Nay - dnay at libertyisp.com <mailto:dnay at libertyisp.com>
> > >
> > >
> > >
> > > ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers at lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
> >
> 
> 
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