Thank you Brian,
 
I ran FS with TPORT_LOG=1 ./freeswitch as you said and I got the output below:
 
Note that 3462101 is the DID provider username so it seems to me that the invitation is going to the right place.
 
In  one part of the output there is-   Processing Unknown->3462101@public - so I think it is trying to reach this username as if  it was a FS registered extension in the public profile, and this is not the case.
 
Am I thinking right ?
 
 
 
freeswitch@maui> recv 995 bytes from udp/[208.239.76.169]:5060 at 19:51:09.809597:
   ------------------------------------------------------------------------
   INVITE sip:3462101@24.67.78.200:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 208.239.76.169:5060;branch=z9hG4bKrsqhv3rnbmpm7tkc71v7va10j6
   CSeq: 1 INVITE
   Call-ID: SDn17k701-7b10910a30ef844aaa5223619da54b89-gurpkk2
   Supported: replaces
   From: Unknown <sip:551133015337@216.143.130.65>;tag=SDn17k701-05eafc8d
   Content-Type: application/sdp
   Allow: INVITE
   X-DID: 3105266066
   X-UUID: d5b75aa4287441c7ba85adcc573d5b6a
   To: <sip:3462101-g471esoujgor0@10.0.5.66:5060;useradd=24.67.78.200;userport=5080;transport=udp>
   Contact: <sip:208.239.76.169:5060;transport=udp;wlsscid=1ae4691c270666;appsessionid=app-13xszycpvbbhq>
   Content-Length: 337
   Max-Forwards: 69
  
   v=0
   o=root 21292 21292 IN IP4 216.143.130.65
   s=session
   c=IN IP4 216.143.130.65
   t=0 0
   m=audio 9758 RTP/AVP 0 8 18 3 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 415 bytes to udp/[208.239.76.169]:5060 at 19:51:09.809999:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 208.239.76.169:5060;branch=z9hG4bKrsqhv3rnbmpm7tkc71v7va10j6
   From: Unknown <sip:551133015337@216.143.130.65>;tag=SDn17k701-05eafc8d
   To: <sip:3462101-g471esoujgor0@10.0.5.66:5060;useradd=24.67.78.200;userport=5080;transport=udp>
   Call-ID: SDn17k701-7b10910a30ef844aaa5223619da54b89-gurpkk2
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
   Content-Length: 0
  
   ------------------------------------------------------------------------
2008-09-15 12:51:09 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/551133015337@216.143.130.65 [356d24c1-37f5-4af6-b911-7d130001a7bd]
2008-09-15 12:51:09 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Unknown->3462101@public
2008-09-15 12:51:09 [WARNING] mod_dialplan_xml.c:252 dialplan_hunt() context public not found
2008-09-15 12:51:09 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
2008-09-15 12:51:09 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/external/551133015337@216.143.130.65 [CS_ROUTING] [NO_ROUTE_DESTINATION]
send 696 bytes to udp/[208.239.76.169]:5060 at 19:51:09.966296:
   ------------------------------------------------------------------------
   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP 208.239.76.169:5060;branch=z9hG4bKrsqhv3rnbmpm7tkc71v7va10j6
   From: Unknown <sip:551133015337@216.143.130.65>;tag=SDn17k701-05eafc8d
   To: <sip:3462101-g471esoujgor0@10.0.5.66:5060;useradd=24.67.78.200;userport=5080;transport=udp>;tag=9XZ4vc3DeNpvp
   Call-ID: SDn17k701-7b10910a30ef844aaa5223619da54b89-gurpkk2
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk
   Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION"
   Content-Length: 0
  
   ------------------------------------------------------------------------
2008-09-15 12:51:09 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 1 (sofia/external/551133015337@216.143.130.65) Ended
2008-09-15 12:51:09 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/551133015337@216.143.130.65 [CS_HANGUP]
recv 418 bytes from udp/[208.239.76.169]:5060 at 19:51:10.069046:
   ------------------------------------------------------------------------
   ACK sip:3462101@24.67.78.200:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 208.239.76.169:5060;branch=z9hG4bKrsqhv3rnbmpm7tkc71v7va10j6
   From: Unknown <sip:551133015337@216.143.130.65>;tag=SDn17k701-05eafc8d
   To: <sip:3462101-g471esoujgor0@10.0.5.66:5060;useradd=24.67.78.200;userport=5080;transport=udp>;tag=9XZ4vc3DeNpvp
   Call-ID: SDn17k701-7b10910a30ef844aaa5223619da54b89-gurpkk2
   CSeq: 1 ACK
   Content-Length: 0
  
   ------------------------------------------------------------------------
 
 

Jair Santos
Software Engineer
Cliconnect Internet Telephony

Brazil: 01155-11-3301-5337
USA: (310)526-6066

 

 

 

-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, September 15, 2008 10:33 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DID not working


On Sep 15, 2008, at 12:26 PM, Jair Santos wrote:

Hello,
 
I am trying to use a DID so I included the following on  dialplan/public.xml
 
 <extension name="inphonex_DID">
      <condition field="destination_number" expression="13105266066">
        <action application="transfer" data="1001 XML default"/>
      </condition>
    </extension>


Chances are the DID is sent into you without the 1 at the beginning.  

Try this

TPORT_LOG=1 ./freeswitch


See what you can tell is going on.  Also "console loglevel debug" and see if maybe it prints out what is going on.


 
 
Here is the sofia status
 
 sofia status
API CALL [sofia(status)] output:
                     Name          Type                               Data      State
=================================================================================================
                 internal       profile   sip:mod_sofia@192.168.1.117:5060      RUNNING (0)
                 external       profile    sip:mod_sofia@24.67.78.200:5080      RUNNING (0)
                 inphonex       gateway      sip:3462101@sip.varphonex.com      REGED
                      nat       profile    sip:mod_sofia@24.67.78.200:5070      RUNNING (0)
                  default         alias                           internal      ALIASED
             voipclic.com         alias                           internal      ALIASED
                 outbound         alias                           external      ALIASED
=================================================================================================
3 profiles 3 aliases
When I call the DID the extension 1001 does not ring.
 
Any help will be very much appreciated.
 
thanks
 
Jair Santos