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@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@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>_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org