[Freeswitch-users] FS installed, but no external calls

[Brian West]

You'll need to set from-user on freeswitch also.

/b

On Jun 4, 2008, at 7:26 PM, Brian B wrote:

> Here are the Elastix/Asterisk settings that were working.  Brian  
> West remarked that I have to "add it to the dialplan in addition to  
> registration".   Can anyone give me a pointer on that?
>
> Inbound
> registration string: 16643:password at did.voip.les.net/16643
> User context: from-trunk
> User Details:
> canreinvite=no
> context=from-trunk
> insecure=very
> nat=yes
> type=user
>
> Outbound
> canreinvite=no
> context=from-trunk
> dtmf=inband
> dtmfmode=inband
> fromuser=16643
> host=did.voip.les.net
> insecure=very
> nat=yes
> qualify=yes
> secret=pass
> type=peer
> user=phone
> username=16643
>
> On Wed, Jun 4, 2008 at 4:08 PM, Brian West <brian at freeswitch.org>  
> wrote:
> how about you reply with the settings you currently use for Asterisk?
>
> /b
>
> On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
>
> > Hi Brian,
> >
> > I created a LES profile located at sip_profiles/external/les.xml
> > with following content:
> >
> > <include>
> >  <gateway name="did.voip.les.net">
> >  <param name="username" value="1490236124"/>
> >  <param name="realm" value="did.voip.les.net"/>
> >  <param name="password" value="mash.n2rown4"/>
> >  </gateway>
> > </include>
> >
> >
> > My dialing is actually done in a Javascript file in the following  
> way:
> >
> > session.execute("bridge","sofia/gateway/did.voip.les.net/
> > 14156113200");
> >
> > Since you already have LES working with Asterisk, I guess you know
> > about the settings that you need on the LES side. But you (or
> > somebody else) might still want to know what is works for me.
> >
> >
> > In the les.net management console I createa a Peer/Trunk.  To do
> > this follow the following steps:
> >
> > 1) Login at les.net
> > 2) On the left menu click on "Peers / Trunk"
> > 3) then click on "create peer"
> > 4) Next click on "Edit" to edit the settings for this trunk.
> > 5) Specify the IP Address and the password of the trunk.
> >
> >
> > Hope this solves your problem.
> >
> > Klaus.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -------- Original-Nachricht --------
> >> Datum: Wed, 4 Jun 2008 12:37:51 -0700
> >> Von: "Brian B" <freeswitch at brian-burt.com>
> >> An: freeswitch-users at lists.freeswitch.org
> >> Betreff: [Freeswitch-users] FS installed, but no external calls
> >
> >> FS is definitely up and running on my Lylix VPS, which is exciting.
> >> X-Lite
> >> is dialed in and can talk to the IVR etc.  But no inbound or  
> outbound
> >> calls.
> >>
> >> I tried configuring les.net (which I got working in * on the same
> >> box) but
> >> couldn't get that working, and sofia status is NOREG.
> >>
> >> I got gizmo to where status is REGED but from XLite when I dial a
> >> 10 digit
> >> number the console shows
> >> [INFO] switch_core_state_machine.c:114
> >> switch_core_standard_on_routing()
> >> No
> >> Route, Aborting
> >>
> >>
> >> Any tips ...or is there an affordable contractor out there who can
> >> get a
> >> "hello world" call going?
> >>
> >>
> >>
> >> sofia status
> >> API CALL [sofia(status)] output:
> >>                     Name          Type
> >> Data      State
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >>  
> =====================================================================
> >>                 internal       profile
> >> sip:mod_sofia at 67.228.218.111:5060
> >> RUNNING (0)
> >>                      nat       profile
> >> sip:mod_sofia at 67.228.218.111:5070
> >> RUNNING (0)
> >>                  default         alias
> >> internal      ALIASED
> >>                 external       profile
> >> sip:mod_sofia at 67.228.218.111:5080
> >> RUNNING (0)
> >>                    gizmo       gateway
> >> sip:17476481805 at proxy01.sipphone.com<sip%3A17476481805 at proxy01.sipphone.com
> >> >
> >> REGED
> >>                   lesnet       gateway
> >> sip:16643 at did.voip.les.net<sip%3A16643 at did.voip.les.net>
> >> NOREG
> >>           67.228.218.111         alias
> >> internal      ALIASED
> >>                 outbound         alias
> >> external      ALIASED
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >> =
> >>  
> =====================================================================
> >>
> >>
> >> Thanks for any tips or suggestions!
> >>
> >> -Brian B
> >
> > --
> > Psssst! Schon vom neuen GMX MultiMessenger gehört?
> > Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080604/098826cc/attachment.html