Yes, the error is "UNALLOCATED NUMBER" which means a phone number that is not in service or possibly a wrong number format. This means that you are actually talking to your provider now which is a good thing. Next what you need to do is ask the provider to analyze what you're sending to see if it is in the correct format.

-MC

On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t@gmail.com> wrote:
Getting a little further, but calls are still failing.
Not sure why something to do with 404 error. could it be wrong number format?

changed dialplan to <action application="bridge" data="sofia/external/01869$1@80.40.150.150:5060"/> just in case the port number was incorrect.



On 2 August 2011 19:16, Michael Collins <msc@freeswitch.org> wrote:
Lloyd,

The gateway you created I think has a few issues. Let's have you bypass it altogether with a single change to your dialplan. Change your bridge line to this:

<action application="bridge" data="sofia/external/01869$1@80.40.150.150"/>

This bypasses the gateway and sends the call straight out the external profile. (You could also send it out the internal profile.)

Try that and see what happens. If you have issues then do the usual console log and siptrace and put it into pastebin.freeswitch.org. Be sure to choose "FreeSWITCH Log" for the syntax highlighting.

-MC


On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t@gmail.com> wrote:
Just did a test, but no joy. I suspect I may have to dispense with the gateway settings and just bridge straight from the dial plan, but it is just a guess.


dialplan
----------------------------------------
<extension name="Local Dial">
   <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
     <action application="bridge" data="sofia/gateway/phisys-2circles/01869$1"/>
   </condition>
</extension>

gateway
---------------------------------------
<include>
  <gateway name="phisys-2circles">
    <param name="from-domain" value="80.40.150.150"/>
    <param name="proxy" value="80.40.150.150"/>  
    <param name="sip-port" value="5060"/>
  </gateway>
</include>



errors
------------------------------------
2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/200@phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42 +0100]
EXECUTE sofia/internal/200@phisys.tele.phi.co.uk bridge(sofia/gateway/phisys-2circles/01869321110)
2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A [CS_NEW]
2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 () Running State Change CS_DESTROY
2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A) State DESTROY
2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A) State DESTROY going to sleep
2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/200@phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup sofia/internal/200@phisys.tele.phi.co.uk [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal sofia/internal/200@phisys.tele.phi.co.uk [KILL]


On 1 August 2011 20:50, Michael Collins <msc@freeswitch.org> wrote:
Okay, so what happens when you dial out? Sorry, it's been a few days and I don't recall where we left off. Be sure to include console log w/ siptrace on pastebin.freeswitch.org.

-MC


On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t@gmail.com> wrote:
I think they have my IP on a white list.


On 1 August 2011 16:24, Michael Collins <msc@freeswitch.org> wrote:
Do they challenge you (digest auth) or do they have your IP address on a white list? That's a critical piece of information that only your provider can supply.

-MC


On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t@gmail.com> wrote:
OK Inbound working with:

  <!--2circles -->  
    <list name="domains" default="allow">
      <node type="allow" cidr="80.40.150.150/32"/>
    </list>

Just need to sort outbound.


On 30 July 2011 04:59, lloyd thomas <lloydie.t@gmail.com> wrote:
Hi, dialling in produces the following error.

2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150 Rejected by acl "domains". Falling back to Digest auth.
2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from ip 80.40.150.150



On 30 July 2011 04:34, lloyd thomas <lloydie.t@gmail.com> wrote:
I am registering with a them. I could not find suitable example in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which


On 29 July 2011 21:57, Michael Collins <msc@freeswitch.org> wrote:
Are you registering with the provider or are they registering with you? If they register with you then a user example is appropriate. If you are registering with them then all you need is a gateway configured. 
-MC


On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t@gmail.com> wrote:
Sorry, example is not clear to me.
I don't understand why a user config is relevant to sip registration for a provider.
An example will help me more. Maybe CIDR attribute in a sip_profile gateway could help.


On 29 July 2011 19:55, Steven Ayre <steveayre@gmail.com> wrote:
Look at the cidr attribute in the user directory to authenticate by IP:
http://wiki.freeswitch.org/wiki/Acl#Users

-Steve

On 29 July 2011 19:38, lloyd thomas <lloydie.t@gmail.com> wrote:

Hi I need a little help setting up a SIP registration for a provider that does not use auth.

All I have is info below.



SBC/Proxy IP:        80.40.150.150:5060

Authentication:      Trusted IP – 88.221.85.33

Assigned DDI:        01869******, 01869******

DTMF Method:         RFC2833

Status:              Live

No. of trunks:       2x

Session Timer:       1800

Profile:             Generic (35060)


Apparently the following is used for *

[vibe]

type = friend

host = 80.40.150.150


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