[Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.

[julien]

Michael Collins a écrit :
>
> A 404 in SIP is just like a 404 when web surfing: the target server 
> can't find whatever it is that you're looking for. In other words, 
> your FS server made contact with the server at the far end, told it 
> what endpoint you're looking for, and the server there said, "I can't 
> find this endpoint, sorry." You'll need to see what's happening at the 
> far end, like whether there really is a 300 or not.
> -MC
>
Thanks for your help Michael.
Thanks to the SIP trace mode, I saw that, as you said, the remote server 
can be contacted but, it can't find the phone with the number 300 :

send 1135 bytes to udp/[PBX IP ADDRESS]:5060 at 07:36:50.305204:
   ------------------------------------------------------------------------
   INVITE sip:300@[PBX IP ADDRESS] SIP/2.0


recv 463 bytes from udp/[FS IP ADDRESS]:5060 at 07:11:33.165472:
   ------------------------------------------------------------------------
   SIP/2.0 404 Not Found
   Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY
   User-Agent: OxO_GW_710/022.001
   To: <sip:300@[PBX IP ADDRESS]>;tag=f75e19901fe0d4a3eed1295e30a93b53

But the private number 300 exist and work very well. So i guess the way 
to join the endpoint (an Alcatel IP phone) is not the good one. I even 
tried with the public number. I don't think the pbx can find 300@[PBX IP 
ADDRESS], it can only find 300 as this phone already has an IP address.