[Freeswitch-users] Configuration

[Jonas Gauffin]

Back to basics. I've reverted to the default sip profile configs.
I've now specified "192.168.1.102" as the domain (the FS ip). Both
phones do now register using that ip.
The registration goes OK.

But when I call from one phone to another (both on the same lan as
FS), I  only get audio one way.
I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan.
This must be the simplest test case, right?

On Jan 8, 2008 2:00 PM, Brian West <brian.west at mac.com> wrote:
> Btw if you put back the config to the default. It will work because
> the ip is the domain and the packet don't loop thur your nat router
> needlessly.
>
> /b
>
> Sent from my iPhone
>
> On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin at gmail.com>
> wrote:
>
>
> >>> * I only get audio one way when dialing from one registered UA to
> >>> another, check attached log internal_call.log
> >>
> >> Chances are mydomain.com is your public IP right? and I suspect NAT
> >> is
> >> involved?  If so then you need to setup a profile for Natted phones
> >> behind the same nat.  One profile for phones outside the nat to
> >> register with FreeSWITCH ... and the public one listed below.
> >>
> >
> > mydomain.com points on my public ip, correct. The phones and FS are
> > behind a router (on the same lan).
> > FS have loaded the three profiles (nat, default and outbound). I have
> > made no changes to them.
> > Currently I have just tested with phones that is on the same lan as
> > FS.
> >
> > here is a registration trans:
> >
> >   REGISTER sip:mydomain.com SIP/2.0
> >   Via: SIP/2.0/UDP
> > 192.168.1.123:
> > 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93
> >   From: <sip:tobbe at mydomain.com>;tag=66055002
> >   To: <sip:tobbe at mydomain.com>
> >   Call-ID: 58ee65fb0b696f5a370d839058387f2c at 192.168.1.123
> >   Contact: <sip:tobbe at 192.168.1.123:5060>
> >   CSeq: 53 REGISTER
> >   Expires: 300
> >   Authorization: Digest username="tobbe",realm="mydomain.com",[...
> > some info stripped out
> > ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5
> >   User-Agent: CMI CM5K
> >   Content-Length: 0
> >
> > tobbe has user_context set to "default" in conf/directory/default/
> > tobbe.xml
> >
> >>
> >>>
> >>>
> >>> * Inbound external calls do not work, i get [407][Proxy
> >>> Authentication Required], check log external_call.log
> >>
> >> This is why you need to start another profile to accept calls from
> >> the
> >> outside world with authentication turned off.  That would fix that
> >> problem.
> >
> > well. the nat profile exists and it has auth_calls set to false. but
> > maybe it isnt used in my config?
> >
> >>
> >> I honestly like having multiple UA's it solves many problems.
> >>
> >
> > what do u mean by that? I'm using multiple UA's, all of them are
> > currently on the same LAN as FS right now (testing enviroment), but
> > later on, they may also be elsewhere.
> > But currently I'm trying to get two UA's to be able to talk with each
> > other (as I said, the audio is just going one way).
> >
> > And I'm not saying that it's one UA to rule them all.. =) I'm just
> > trying to understand this all new configuration how a profile is
> > selected (for inbound/outbound external calls, internal calls,
> > internal registration (on the same lan as FS) and external
> > registration (not on the same lan as FS)).
> >
> > thanks,
> >  jonas
> >
>
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