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 ? <mailto:freeswitch@maui> 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=a pp-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 <mailto:sofia/external/551133015337@216.143.130.65> 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 <mailto:sofia/external/551133015337@216.143.130.65> 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 ( <mailto:sofia/external/551133015337@216.143.130.65> 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 <mailto:sofia/external/551133015337@216.143.130.65> 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 <http://www.cliconnect.com/> Cliconnect Internet Telephony <http://maps.google.com/maps?q=&hl=en> <http://www.linkedin.com/img/signature/pic_plastic_cool_26x130.gif> 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
