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
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