[Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
Filip Lyncker
lyncker at lyth.de
Tue Sep 22 07:29:29 PDT 2009
now i registered from my x-lite client without anyproblems.
but I think i got it now, my tcpdump says the following :
IP 192.168.1.119.5060 > 93.210.212.xxx.5080: SIP, length: 465
wich is the external IP of my network ! must have somthing todo with NAT
/ Masquerade options... how can I avoid this ?
thanks for your help ...
regards,
filip
Tihomir Culjaga schrieb:
> hmmm .. can you register using x-lite or some other softphone with the
> same credentials?
>
> can you paste a siptrace of the failed registration?
>
>
> BTW: Make sure nothing is already registered with this credentials
> when you try with FS
>
> T.
>
> On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker <lyncker at lyth.de
> <mailto:lyncker at lyth.de>> wrote:
>
> 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>
> > <mailto: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>
> <mailto: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>
> <mailto: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>
> <mailto: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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> > FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> >
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> >
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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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--
_________________________________
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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