[Freeswitch-users] FS and Asterisk connectivity

[Daniel Hefti]

I actually was trying this out just recently, and I got the same results: inbound calls aren't able to come in for asterisk.  (I wasn't able to get the debug output, though... did you have to re-compile sofia after exporting the debug variables to get the debug info?)

Anyways, here's what I did:

I did a bind to the address 127.0.0.2 in sip.conf, created the virtual network adapter with the ip 127.0.0.2, fired up asterisk, used similar configurations you used, but set the realm to 127.0.0.2, registration to true, and setup in the dialplan a means of calling out.  Then I fired up freeswitch.  Freeswitch registers with asterisk, and any attempt I make to dial out works.

Initially I noticed when calls come in from asterisk, freeswitch started complaining that there was no user named the same name as the username used in asterisk, so I created a user with the same name and password in the dialplan.  Now, I get no useful output from freeswitch, and asterisk complains that there's no route to destination.

I also tried setting up the same user account on a softphone, which registers and accepts calls locally from other sip phones, but not from asterisk through freeswitch.

As a last ditch effort, I made a small extension in the dialplan to send all calls that match condition:
        <condition field="destination_number" expression=".*">
to a given phone, which works for all requests I've made except the ones from asterisk.  So that didn't work either, and I've run out of ideas.

-Dan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Snipes
Sent: Tuesday, April 15, 2008 3:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS and Asterisk connectivity


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.

Thanks,
Brian


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