hi Filip,<br><br><br>for calling a user... please read this first: <a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User</a><br>
for making a GW register into e.g. asterisk please use this:<br><br><br><include><br> <gateway name="gw01"><br> <param name="username" value="USERNAME_ON_ASTERISK"/><br> <param name="realm" value="ASTERISK_IP_ADDRESS"/><br>
<param name="password" value="PASSWORD_ON_ASTERISK"/><br> <param name="register" value="true"/><br> <param name="caller-id-in-from" value="true"/><br>
</gateway><br></include><br><br>this should be enough to register the GW... after that please read this: <a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways</a><br>
<br><br>in your case it will be something like this:<br><br><extension name="dialGW"><br> <condition field="destination_number" expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"><br> <action application="bridge" data="sofia/gateway/gw01/$1"/><br>
</condition><br></extension><br><br><br><br><br><br><br><br><br><br><div class="gmail_quote">On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <span dir="ltr"><<a href="mailto:lyncker@lyth.de">lyncker@lyth.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi List,<br>
<br>
for the first experiments with freeswitch I downloaded the Windows<br>
installation.<br>
Now Im trying to get my 2 Sipphones get connected to. Later I want<br>
connect the freeswitch to my asterisk gateway.<br>
<br>
I find the examples pretty complex therfore Im trying to build up a<br>
simple solution to understand the functions from the scratch ..<br>
<br>
my current problem is , that I cant route my local sips to each other (<br>
registration seems to work now).<br>
the next is , that freeshwitch is not able to connect to asterisk. but I<br>
will describe this later.<br>
<br>
I installed in the Directory a xml file ( called 22.xml) with the<br>
following content :<br>
<br>
<include><br>
<domain name="$${domain}"><br>
<user id="22" mailbox="22"><br>
<params><br>
<param name="password" value="Xk21%"></param><br>
<param name="vm-password" value="22"></param><br>
<param name="sip-port" value="5060"></param><br>
<br>
</params><br>
<variables><br>
<variable name="accountcode" value="22"></variable><br>
<variable name="user_context" value="default"></variable><br>
<variable name="effective_caller_id_name" value="Extension<br>
22"></variable><br>
<variable name="effective_caller_id_number" value="22"></variable><br>
</variables><br>
</user><br>
<user id="24" mailbox="24"><br>
<params><br>
<param name="password" value="dudeldum"></param><br>
<param name="vm-password" value="24"></param><br>
<param name="sip-port" value="5060"></param><br>
<br>
</params><br>
<variables><br>
<variable name="accountcode" value="24"></variable><br>
<variable name="user_context" value="default"></variable><br>
<variable name="effective_caller_id_name" value="Extension<br>
24"></variable><br>
<variable name="effective_caller_id_number" value="24"></variable><br>
</variables><br>
</user><br>
</domain><br>
</include><br>
<br>
This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I<br>
configured this dialplan :<br>
<br>
<include><br>
<context name="any"><br>
<condition field="destination_number" expression="^(2[0-9])$"><br>
<br>
<action application="bridge" data="user/${dialed_extension}"/><br>
<br>
</condition><br>
</include><br>
<br>
wich doesnt work , mybe b/c the user/${dialed_extension} I dont know...<br>
Freeswitch says:<br>
[INFO] switch_core_state_machine.c:136 No Route, Aborting<br>
[NOTICE] switch_core_state_machine.c:137 Hangup<br>
sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> [CS_ROUTING] [NO_ROUTE_DESTINATION]<br>
[NOTICE] switch_core_session.c:1086 Session 17<br>
(sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a>) Ended<br>
[NOTICE] switch_core_session.c:1088 Close Channel<br>
sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> [CS_DESTROY]<br>
<br>
Im sure , for you guys this cant be a big deal;)<br>
<br>
<br>
Next Point is my Asterisk registration , mybe you can help me out here<br>
to .. :<br>
<br>
In the sip-profiles/external I installed the my_asterisk.xml with that<br>
content :<br>
<br>
<include><br>
<gateway name="asterisk"><br>
<param name="username" value="28"></param><br>
<param name="password" value="test"></param><br>
<param name="realm" value="28"></param><br>
<param name="proxy" value="192.168.1.119"></param><br>
<param name="register" value="true"></param><br>
</gateway><br>
</include><br>
<br>
Freeswitch allways complains a timeout error :<br>
[ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request<br>
Timeout [408]. failure #17<br>
[WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry<br>
to 540 seconds.<br>
<br>
it seems that It cant connect ( I also tried out to explicit set the<br>
port to 5060 b/c I read something about 5080 .. : <param name="sip-port"<br>
value="5060"></param> but this didnt help)<br>
In my Asterisk I set in the sip.conf the entry 28 with the pw test ....<br>
<br>
<br>
If someone could help me with my first steps I would be verrry thankful ;))<br>
<br>
cheers<br>
<br>
<br>
Filip<br>
<br>
--<br>
_________________________________<br>
Filip Lyncker, Dipl.-Inform. (FH)<br>
<br>
<br>
Lyncker & Theis GmbH<br>
Wilhelmstr. 16<br>
65185 Wiesbaden<br>
Germany<br>
<br>
Fon +49 611/9006951<br>
Fax +49 611/9406125<br>
<br>
<br>
Handelsregister: HRB 23156 Amtsgericht Wiesbaden<br>
Steuernummer: 4023897051<br>
USt-IdNr.: DE255806399<br>
<br>
Geschäftsführer:<br>
Filip Lyncker,<br>
Armin Theis<br>
<br>
<br>
<br>
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</blockquote></div><br>