[Freeswitch-users] FS and Asterisk connectivity

Thu Apr 24 16:15:59 PDT 2008

Asterisk was spewing out tons of funky lines with those two enabled, so I disabled them.  Thanks for pointing that out, Snipes.

As for the pastbin posts, it doesn't look like their available anymore...

I think managed to get a little further.

I moved the gateway to the outbound folder instead, as West pointed out.  (It is not available in default's sip_profiles directory anymore, and there is no announcements of it being supplied twice.)  I then set the sip peer in asterisk to a static address... using the machine's local ip address; so no registration necessary.

Afterwards I still kept getting a 'can't find user' warning from sofia...

[WARNING] sofia_reg.c:935 sofia_reg_parse_auth() can't find user [freeswitch at xxx.xxx.xxx.xxx]

...while asterisk showed a return response, "Forbidden".  I noticed that that when sofia loads, the same ip address (stated above as x's) is aliased:

[NOTICE] sofia.c:1461 config_sofia() Adding Alias [xxx.xxx.xxx.xxx] for profile [default]

I don't know if that's related, but when I re-added a user called freeswitch to the default's directory, the message disappears. Then asterisk received the same response...

Long story short.  I believe the problem was that in the freeswitch user's profile I defined in the directory had a password parameter.  When I commented that out and set the user_context to public, I was able to get the demo ivr (like it's supposed to in the public.xml dialplan).

All these defaults can drive a person crazy.  Is /conf/directory/default and its related default.xml in any way related to sofia specifically, to the dialplan, or primarily to FreeSwitch as a whole?  My original idea was that the directory was primarily a subsection of related stuff with respect to sofia, but now I'm beginning to see it as a larger whole, whose parameters and variables can be or are typically related to other modules, such as how the user_context primarily relates how the user should traverse the dialplan, but users defined in it are independent of such modules.  The folder, 'default', is just provided for convenience.  I guess I'm just trying to formulate how it all comes together.  After working with gateways, I think I'm beginning to get it.

Coolness.  Forgive my rantings.  Off to do some further testing.

-Dan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Hefti
Sent: Thursday, April 24, 2008 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Re: [Freeswitch-users] FS and Asterisk connectivity

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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