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 email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com 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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