[Freeswitch-users] Some Newbie questions about dialplan and local Sip registration

Filip Lyncker lyncker at lyth.de
Tue Sep 22 03:56:39 PDT 2009


Hi Tihomir,

Thanks for your help , I added the Asteriskparameters as you described 
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed 
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration 
Failed with status Request Timeout [408]. failure #9

Now, my gateway entry looks like the following :

<include>
  <gateway name="asterisk">
  <param name="username" value="28"/>
  <param name="realm" value="192.168.1.119"/>  
  <param name="proxy" value="192.168.1.119"/>
  <param name="password" value="test"/>
  <param name="register" value="true"/>
  <param name="caller-id-in-from" value="true"/> 
  <param name="sip-port" value="5060"></param>
  </gateway>
</include>


What can be still wrong here?

Regards,

Filip



Tihomir Culjaga schrieb:
> 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 <lyncker at lyth.de 
> <mailto:lyncker at 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 at 192.168.1.34 <mailto:24 at 192.168.1.34>
>     [CS_ROUTING] [NO_ROUTE_DESTINATION]
>     [NOTICE] switch_core_session.c:1086 Session 17
>     (sofia/internal/24 at 192.168.1.34 <mailto:24 at 192.168.1.34>) Ended
>     [NOTICE] switch_core_session.c:1088 Close Channel
>     sofia/internal/24 at 192.168.1.34 <mailto:24 at 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
>
>
>
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-- 
_________________________________
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 






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