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@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@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@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@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@64.34.181.47 [CS_HANGUP] On Wed, Jun 4, 2008 at 5:36 PM, Brian West <brian@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@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@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@brian-burt.com> An: freeswitch-users@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@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com> <sip%3A17476481805@proxy01.sipphone.com<sip%253A17476481805@proxy01.sipphone.com>
REGED lesnet gateway sip:16643@did.voip.les.net <sip%3A16643@did.voip.les.net>< sip%3A16643@did.voip.les.net <sip%253A16643@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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