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