False,

I think Brian is sleeping so I give you a quick answer.

For OUTBOUND:

Following my previous mail, you have to check if your user 1000 has the default context configured.
If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
(You may also remove from this path the remaining .xml files from the default install)
In 01_whatever.xml, put:

<include>
    <extension name="Route_to_sip.yeah.com">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="bridge" data="sofia/gateway/sip.yeah.com/0$1"/>
      </condition>
    </extension>
<include>

So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.

For INBOUND:

In conf/dialplan/public/, add a file named whatever.xml with the following content:

<include>
   <extension name="Inbound_from_sip.yeah.com">
      <condition field="destination_number" expression="^YOUR_DID$">
        <action application="transfer" data="1000 XML default"/>
      </condition>
    </extension>
</include>

Here, YOUR_DID is the DID number your provider allocated to you.
That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.

Of course, after doing all this, reloadxml.

It will perhaps not fit in your config exactly, but you should get the idea to get started.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

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Le 30/03/2010 à 09:01, false a écrit :

Hi Brian

could you guide me more to config on freeswitch

i want xilte with user 1000 can make call out to sip server
and the prefix when make call out to sip server : 0
and receive incomming call from SIP server

what file name i should create

Thank you


--- Ngày Thứ 3, 30/03/10, Brian West <brian@freeswitch.org> đã viết:

Từ: Brian West <brian@freeswitch.org>
Chủ đề: Re: [Freeswitch-users] need help on call to gateway
Đến: freeswitch-users@lists.freeswitch.org
Ngày: Thứ Ba, 30 tháng 3, 2010, 4:37

its looking in context default... not public.

/b

On Mar 29, 2010, at 11:33 PM, false wrote:

Hi all

i use freeswitch to register to sip server
the domain is not real: sip.yeah.com
the outbound proxy : 118.69.145.5

i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
</include>
    <gateway name="sip.yeah.com">
        <param name="username" value="071234"/>
        <param name="from-user" value="071234"/>   
        <param name="proxy" value="sip.yeah.com"/>
        <param name="username" value="071234"/>
        <param name="password" value="1234"/>
        <param name="register-proxy" value="118.69.145.5"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="10"/>
    </gateway>
</include>

so i use xlite register to freeswitch with user 1000
i start freeswitch , freeswitch start ok,
freeswitch register to sip server successfull with username 071234
xlite register to freeswitch successfull

so i edit the public.xml
for incomming call from sip server, i will forward to extension 1000
    <extension name="sip.yeah.com">
      <condition field="destination_number" expression="071234">
        <action application="transfer" data="1000 XML default"/>
      </condition>
    </extension>

but when i use xlite make call out from freeswitch to sip server and get no routing


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