False,

The logic in FS is that a call will go through the dialplan context attached to the SIP profile that the call is coming into.
So if you have the SIP Profile internal bound to 10.0.0.1:5080 for instance, and then you receive an INVITE on 10.0.0.1:5080, FS will parse the dialplan context configured in internal.xml (by default, public).

For registered users though, you can alter this behavior by specifying a context in the user variables:

      <variable name="user_context" value="your_own_context_for_this_user"/>

If you don't, the context configured in the SIP profile that the user is sending INVITEs to will be used.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




Le 30/03/2010 à 06:33, false a écrit :

Hi all

i use freeswitch to register to sip server
the domain is not real: sip.yeah.com
the outbound proxy : 118.69.145.5

i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
</include>
    <gateway name="sip.yeah.com">
        <param name="username" value="071234"/>
        <param name="from-user" value="071234"/>   
        <param name="proxy" value="sip.yeah.com"/>
        <param name="username" value="071234"/>
        <param name="password" value="1234"/>
        <param name="register-proxy" value="118.69.145.5"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="10"/>
    </gateway>
</include>

so i use xlite register to freeswitch with user 1000
i start freeswitch , freeswitch start ok,
freeswitch register to sip server successfull with username 071234
xlite register to freeswitch successfull

so i edit the public.xml
for incomming call from sip server, i will forward to extension 1000
    <extension name="sip.yeah.com">
      <condition field="destination_number" expression="071234">
        <action application="transfer" data="1000 XML default"/>
      </condition>
    </extension>

but when i use xlite make call out from freeswitch to sip server and get no routing

nta: received INVITE sip:071234@192.168.1.90;user=phone SIP/2.0 (CSeq 3621)
nta: INVITE (3621) going to a default leg
nua(0xb6e20f90): adding session usage
nta: sent 100 Trying for INVITE (3621)
nua(0xb6e20f90): event i_invite 100 Trying
nua(0xb6e20f90): call state changed: init -> received, received offer
nua(0xb6e20f90): event i_state 100 Trying
432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000@192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006]
446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default
455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000@192.168.1.90 to enum[071234@default]
566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting
566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000@192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION]
nua(0xb6e20f90): recv signal r_respond 404 Not Found

i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch

please help,

Thank you


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