I'll suspect you're trying to challenge the inbound call from Asterisk to FreeSWITCH? If thats the case that would be your problem. The best corse of action is to use the outbound profile to register with Asterisk then put the routing in dialplan/public.xml so we don't try to auth the call inbound from Asterisk. That should fix that. /b On Apr 24, 2008, at 9:59 AM, Daniel Hefti wrote:
I actually was trying this out just recently, and I got the same results: inbound calls aren't able to come in for asterisk. (I wasn't able to get the debug output, though... did you have to re- compile sofia after exporting the debug variables to get the debug info?)
Anyways, here's what I did:
I did a bind to the address 127.0.0.2 in sip.conf, created the virtual network adapter with the ip 127.0.0.2, fired up asterisk, used similar configurations you used, but set the realm to 127.0.0.2, registration to true, and setup in the dialplan a means of calling out. Then I fired up freeswitch. Freeswitch registers with asterisk, and any attempt I make to dial out works.
Initially I noticed when calls come in from asterisk, freeswitch started complaining that there was no user named the same name as the username used in asterisk, so I created a user with the same name and password in the dialplan. Now, I get no useful output from freeswitch, and asterisk complains that there's no route to destination.
I also tried setting up the same user account on a softphone, which registers and accepts calls locally from other sip phones, but not from asterisk through freeswitch.
As a last ditch effort, I made a small extension in the dialplan to send all calls that match condition: <condition field="destination_number" expression=".*"> to a given phone, which works for all requests I've made except the ones from asterisk. So that didn't work either, and I've run out of ideas.
-Dan
Brian West sip:brian@freeswitch.org