[Freeswitch-users] FS installed, but no external calls
Peder @ NetworkOblivion
peder at networkoblivion.com
Thu Jun 5 15:32:45 EDT 2008
In public.xml, you want something like this:
<extension name="test_did">
<condition field="destination_number" expression="^(4153084258)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
and then in default.xml have something like this in the default context:
<extension name="Local_Extension">
<condition field="destination_number" expression="^(4153084258)$"
continue=
"on-true">
<action application="set" data="dialed_ext=$1"/>
</condition>
<condition field="destination_number"
expression="^${caller_id_number}$">
<action application="set"
data="voicemail_authorized=${sip_authorized}"/
>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default $${domain}
${dialed_
ext}"/>
<anti-action application="ring_ready"/>
<anti-action application="set" data="call_timeout=10"/>
<anti-action application="set" data="hangup_after_bridge=true"/>
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="bridge"
data="USER/${dialed_ext}@$${domain}"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default $${domain}
${dialed_e
xt}"/>
</condition>
</extension>
Brian B wrote:
> 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 <mailto: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 <mailto: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
> <mailto: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 <mailto:4153084258 at 64.34.181.47>
> [CS_HANGUP]
>
>
>
>
> On Wed, Jun 4, 2008 at 5:36 PM, Brian West <brian at freeswitch.org
> <mailto: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
>> <http://16643:password@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 <http://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
>> <mailto: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 <http://did.voip.les.net>">
>> > <param name="username" value="1490236124"/>
>> > <param name="realm" value="did.voip.les.net
>> <http://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/
>> <http://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 <http://les.net> management console I createa
>> a Peer/Trunk. To do
>> > this follow the following steps:
>> >
>> > 1) Login at les.net <http://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
>> <mailto:freeswitch at brian-burt.com>>
>> >> An: freeswitch-users at lists.freeswitch.org
>> <mailto: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 <http://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
>> <http://sip:mod_sofia@67.228.218.111:5060>
>> >> RUNNING (0)
>> >> nat profile
>> >> sip:mod_sofia at 67.228.218.111:5070
>> <http://sip:mod_sofia@67.228.218.111:5070>
>> >> RUNNING (0)
>> >> default alias
>> >> internal ALIASED
>> >> external profile
>> >> sip:mod_sofia at 67.228.218.111:5080
>> <http://sip:mod_sofia@67.228.218.111:5080>
>> >> RUNNING (0)
>> >> gizmo gateway
>> >> sip:17476481805 at proxy01.sipphone.com
>> <mailto:sip%3A17476481805 at proxy01.sipphone.com><sip%3A17476481805 at proxy01.sipphone.com
>> <mailto:sip%253A17476481805 at proxy01.sipphone.com>
>> >> >
>> >> REGED
>> >> lesnet gateway
>> >> sip:16643 at did.voip.les.net
>> <mailto:sip%3A16643 at did.voip.les.net><sip%3A16643 at did.voip.les.net
>> <mailto:sip%253A16643 at did.voip.les.net>>
>> >> NOREG
>> >> 67.228.218.111 <http://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
>> <mailto:Freeswitch-users at lists.freeswitch.org>
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>> >
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>>
>>
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