[Freeswitch-users] hi there

Josip Djuricic josip.djuricic at primatel.hr
Mon Mar 10 09:34:07 PDT 2008


Thank you for the info now it works, but I have one other problem.

When I call from voip phone to the outside network the call works just 
fine, but when I call from outside network to the voip phone (connected 
to freeswitch) I can answer the phone, but there is no rtp between, on 
neither side.

Wireshark and Cisco debug ccsip shows that freeswitch returns SIP:415 
Unsupported media type

Anynone having any clues?

I can include lot more debugs if needed?


Brian West wrote:
> You want to send calls to port 5080.  Thats the bottom line since you 
> have phones registering you want to keep that.
>
> sip_profiles/default.xml (for phones to register with NOT gateways to 
> send calls to, runs on port 5060)
> sip_profiles/outbound.xml (all in/out interaction with registered 
> gateways and anonymous inbound sip, runs on 5080)
>
>
> outbound.xml is setup to hit the public context (dialplan/public.xml)
>
> Their are examples in there on how to route a did.
>
> /b
> PS: The Default config are NOT ment to be the only way to configure 
> and run your system.  Its a general config setup to demo how to setup 
> FreeSWITCH. 
>
>
>
> On Mar 10, 2008, at 9:37 AM, Josip Djuricic wrote:
>
>> Thank you Brian,
>>
>> would it be possible maybe to go little more deeper explaining more 
>> in depth these two profiles?
>>
>> Brian West wrote:
>>> You need to understand why and how this works:
>>>
>>> sip_profiles/outbound.xml vs sip_profiles/default.xml
>>>
>>> If you have phones registering then you're better off sending calls to  
>>> port 5080 into the public context as that profile doesn't auth.
>>>
>>> If you really wanna do this then open up default.xml sip profile then  
>>> set auth-calls=false then change the context= to default.
>>>
>>> That should get you going but understand that opens you wide up.
>>>
>>> /b
>>>
>>>
>>> On Mar 10, 2008, at 7:48 AM, Josip Djuricic wrote:
>>>
>>>   
>>>> Hello,
>>>>
>>>> I'm testing freeswitch with AS5300 for inbound and outbound call's.
>>>>
>>>> Does anyone have any info on how to disable Proxy authentication,
>>>> because when the AS5300 sends the call to freeswitch, debug on AS  
>>>> said:
>>>> Received: SIP/2.0 407 Proxy Authentication Required.
>>>>
>>>> I tried to turn off gateway authorization, and setting authless
>>>> authorization, but it just doesn't work.
>>>>
>>>>
>>>> Thanks
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>     
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>   
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org 
>> <mailto:Freeswitch-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080310/bb826af5/attachment-0002.html 


More information about the FreeSWITCH-users mailing list