[Freeswitch-users] FS installed, but no external calls
Brian B
freeswitch at brian-burt.com
Thu Jun 5 11:00:52 PDT 2008
Now the console is showing that there's an inbound call, but I don't know
how to route it to an extension, which I'm gathering is in the dialplan?
There were some "advanced" examples in the documentation but nothing with
something simple like "route all calls to extension 1001" - which I will
plan to add to the wiki when I get this figured out!
I admit to being a complete nube - barely know linux, regex etc. My
contribution will be in documentation for nubes like myself! :-)
Here's the console log:
2008-06-05 01:07:04 [NOTICE] switch_channel.c:533 switch_channel_set_name()
New Channel sofia/external/4153084258 at 64.34.181.47[38a9fd64-dd73-4b6a-9cd9-9cd4675261ea]
# that is my inbound DID
2008-06-05 01:07:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing
NOT FOUND->4153581872 at public
2008-06-05 01:07:04 [INFO] switch_core_state_machine.c:114
switch_core_standard_on_routing() No Route, Aborting
2008-06-05 01:07:04 [NOTICE] switch_core_state_machine.c:115
switch_core_standard_on_routing() Hangup sofia/external/
4153084258 at 64.34.181.47 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2008-06-05 01:07:04 [NOTICE] switch_core_session.c:753
switch_core_session_thread() Session 8 (sofia/external/
4153084258 at 64.34.181.47) Ended
2008-06-05 01:07:04 [NOTICE] switch_core_session.c:755
switch_core_session_thread() Close Channel sofia/external/
4153084258 at 64.34.181.47 [CS_HANGUP]
On Wed, Jun 4, 2008 at 5:36 PM, Brian West <brian at freeswitch.org> wrote:
> 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>
>> <sip%3A17476481805 at proxy01.sipphone.com<sip%253A17476481805 at proxy01.sipphone.com>
>> >> >
>> >> REGED
>> >> lesnet gateway
>> >> sip:16643 at did.voip.les.net <sip%3A16643 at did.voip.les.net><
>> sip%3A16643 at did.voip.les.net <sip%253A16643 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
>> >
>> > _______________________________________________
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>>
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>
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