[Asterisk-Users] One way audio

[Rich Adamson]

Inline and somewhat confused....

> >[...]
> >
> > (Given you mentioned g729, its
> >likely you're trying to use sip across dsl broadband.)
> >  
> >
> Yes
> 
> >If you have a nat/firewall function between * and the sip phone,
> >look there first.
> >
> No nat/firewall between
> 
> > In followup posts, you might mention how your stuff
> >is configured and include appropriate portions of your sip.conf.
> >  
> >
> Setup is
> 
> <--------- SIP phones -------->-*
> <------- SIP softphones ------> | --- asterisk --- internet
> <------- IAX softphones ------> |  on nat/firewall
> <H323 softphones through GnuGK>-*       box

Is the above "on nat/firewall" supposed to say "no nat/firewall"?

So, what that drawing suggests is that asterisk is routing packets
between the Internet and the 192.168.10.x network. Is there something
else involved? If not, then asterisk apparently is your nat box.

> What I face is that a SIP call to our GW has from time to time the 
> behaviour to "loose" audio. Hanging up and retrying can work, but mostly 
> we wait or use an IAX GW and try again and then it work. Can also take 
> few hours before it work again.

Do your log records tend to suggest the IAX link is dropping, a SIP
link to somewhere, or what?
 
> Our provider see the calls like connected with the right codec, and I 
> can see them too in * logs.

If I recall your original posting, it was oriented around a sip call
using g729 dropping connections. How does that relate to the above
diagram and the sip.conf entry below?
 
> This is the sip.conf relevant part from mainphone BT ATA 186 In this 
> one, prefered codec are G729, Ulaw and Alaw.
> 
> [106]
> type=friend                    ; either "friend" (peer+user), "peer" or 
> "user"
> context=default
> qualify=yes
> username=106                   ; usually matches the [section] title
> secret=<secret>
> host=192.168.10.6
> canreinvite=no                 ; allow RTP voice traffic to bypass Asterisk
> dtmfmode=rfc2833               ; either RFC2833 or INFO for the BudgeTone
> disallow=all                   ; need to disallow=all before we can use 
> allow=
> allow=ulaw                     ; Note: In user sections the order of codecs
> allow=alaw                     ; listed with allow= does NOT matter!
> allow=g729                     ; Pass-thru only unless g729 license obtained
> allow=gsm                      ; allways allow gsm
> allow=ilbc

Why is there a "allow=" with nothing after the "=" in the above.

The "canreinvite=no" comment says "allow RTP" which is backwards. What
are you actually expecting?

Since the explanations for the above stuff is a little thin, I'd have
to guess that you might have one or more internal sip phones that don't
have "canreinvite=no" and at least one Internet sip phone that is
unknown as to how it is configured. That's reading way between the lines
and guessing a lot.