[Freeswitch-users] FS and Asterisk connectivity

Brian Snipes bsnipes at snipes.org
Tue Apr 15 13:48:19 PDT 2008

On Tue, 2008-04-15 at 09:34 -0500, Brian Snipes wrote:
> On Mon, 2008-04-14 at 22:10 -0300, Arnaldo de Moraes Pereira wrote:
> > On Mon, Apr 14, 2008 at 4:49 PM, Brian Snipes <bsnipes at snipes.org> wrote:
> > > I wish to connect FS to * for interoperability testing and can't seem to
> > >  get my configs correct.  Has anyone done this already and if so can you
> > >  post your configs?
> > 
> > My configs are like yours, except for three things:
> >   1. my asterisk is configured on outbound profile, instead of default's
> >   2. FS registers to my asterisk
> >   3. My rev: 8081
> > 
> > Are you sure the sampling rate are ok for both legs ? Besides that and
> > comfortable noise generation (turned off, as: <param
> > name="supress-cng" value="true"/>), I can't think of anything else.
> > 
> > My 8081 rev is working nicely with asterisk, maybe I should update and
> > see what happens.
> Thanks for the response Arnaldo.  I just moved it to the outbound
> profile and set the 'suppress-cng' to be uncommented and set to true.
> On the sampling rate, do you mean the codecs in use by the phone and
> allowed on asterisk in the sip.conf file?  Would you mind posting an
> example of your asterisk settings? I guess I have something wrong there
> also.

I was able to get fs to connect to asterisk for outbound calls by
kepping the asterisk.xml in the default profile. When I put it in
outbound it would send the outside ip as the connection number to which
asterisk would send the packets.  Since the fs and asterisk are on the
same lan that is not what I wanted.
The audio issue turned out to be a problem with the snom phones and the
encrypted rtp bug.  Disabling that allowed audio to function.  There are
still several things I need to get figured out:

1. calls from fs to asterisk as a gateway need to have the users name
+number.  It has the name but the number is showing as 'freeswitch'
which is what is in my sip.conf.

2. calls from asterisk to my fs box. I get: [DEBUG] sofia.c:219
sofia_event_callback() event [nua_i_state] status [407][Proxy
Authentication Required] session: n/a.  I am not sure what to change to
allow calls to flow this way.


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