[Freeswitch-users] Configuration

freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org
Tue Jan 8 03:09:57 PST 2008


This is why the default config has three profiles.  In various cases  
of nat involved you have to do some special things to allow it to  
work.  Checkout outbound.xml and nat.xml.  As for audio you need two  
profiles.. one for the communicating with phones behind the same nat  
and one for communicating with the outside world from behind nat.  You  
can't just get away with using one profile for everything in these  
cases.

/b

On Jan 8, 2008, at 5:13 AM, freeswitch-users at lists.freeswitch.org wrote:

> Hello
>
> I still got problems with the configuration, since the new layout  
> was added.
>
> Today I've updated to latest trunk (config files too) and I've  
> configured the system.
>
> I've only done the following changes:
> 1. I've set domain=mydomain.com in vars.xml
> 2. I've renamed sip_profiles/default to mydomain.com and nat to  
> nat_mydomain.com
> 3. I've enabled mod_xml_curl and configured it to fetch dial plans
> 4. I've commented out mod_dialplan_asterisk and mod_iax
> 5. I've added my extensions to directory\default (copied brian's  
> config and made appropriate changes)
>
> My sip gateway do not use authentication (and I therefore just use " number at mygateway.com 
> " in the dialplan for external calls).
> The phones register by using mydomain.com as proxy/registrar (not by  
> using an ip).
>
> The UA's can register fine and external outbound calls works gr8.
>
> Current problems:
>
> * I only get audio one way when dialing from one registered UA to  
> another, check attached log internal_call.log
>
> * Inbound external calls do not work, i get [407][Proxy  
> Authentication Required], check log external_call.log
>
> //Jonas
> < 
> internal_call 
> .log 
> ><external_call.log>_______________________________________________
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