hi Filip,
for calling a user... please read this first: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User 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_gateways 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 <lync...@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/2...@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] > [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/internal/2...@192.168.1.34) Ended > [NOTICE] switch_core_session.c:1088 Close Channel > sofia/internal/2...@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 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Geschäftsführer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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