[Freeswitch-users] DID not working
Jair Santos
jairds at shaw.ca
Mon Sep 15 13:03:47 PDT 2008
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 at 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 at maui> freeswitch at maui> recv 995 bytes from
udp/[208.239.76.169]:5060 at 19:51:09.809597:
------------------------------------------------------------------------
INVITE sip:3462101 at 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 at 216.143.130.65>;tag=SDn17k701-05eafc8d
Content-Type: application/sdp
Allow: INVITE
X-DID: 3105266066
X-UUID: d5b75aa4287441c7ba85adcc573d5b6a
To:
<sip:3462101-g471esoujgor0 at 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 at 216.143.130.65>;tag=SDn17k701-05eafc8d
To:
<sip:3462101-g471esoujgor0 at 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 at 216.143.130.65>
sofia/external/551133015337 at 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 at 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 at 216.143.130.65>
sofia/external/551133015337 at 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 at 216.143.130.65>;tag=SDn17k701-05eafc8d
To:
<sip:3462101-g471esoujgor0 at 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 at 216.143.130.65>
sofia/external/551133015337 at 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 at 216.143.130.65>
sofia/external/551133015337 at 216.143.130.65 [CS_HANGUP]
recv 418 bytes from udp/[208.239.76.169]:5060 at 19:51:10.069046:
------------------------------------------------------------------------
ACK sip:3462101 at 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 at 216.143.130.65>;tag=SDn17k701-05eafc8d
To:
<sip:3462101-g471esoujgor0 at 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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, September 15, 2008 10:33 AM
To: freeswitch-users at 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 at 192.168.1.117:5060
RUNNING (0)
external profile sip:mod_sofia at 24.67.78.200:5080
RUNNING (0)
inphonex gateway sip:3462101 at sip.varphonex.com
REGED
nat profile sip:mod_sofia at 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
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