Rob Emanuele [EMAIL PROTECTED] wrote: > I've got one of those cool free incoming IPKall phone numbers from > www.ipkall.com. These numbers just connect to the SIP proxy of your > choice, they default to Frreworld Dialup. You can use them with your own > sip proxy on asterisk. My config for this is below. > > The trouble I'm having is the incoming calls do not seem to hit the > section in sip.conf for the call. With sip debugging turned on I see the > call come in and the message below is printed. > > If I put the exten route that I have in the ipkall-inbound section of > extensions.conf (below) into the default section it works fine, but isn't > neat and elegant. > > How do I make incoming call from ipkall match a sip.conf section? > > > From sip.conf: > > [3501] > type=peer > host=dynamic > dtmfmode=rfc2833 > context=ipkall-inbound > insecure=very > nat=no > > > From extensions.conf: > > [ipkall-inbound] > exten = 3501,1,Goto(menu,s,1) > You'll probably find that there's no need to set up a specific user for IPKall. You were using "type = peer", which would have been wrong anyway.
In your [general] section, create a "context = incoming-sip" (or whatever you want to call it) and then set up a matching context in extensions.conf. Your extensions.conf context can then match your 3501 extension, along with any other direct incoming SIP addresses you need. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
