> 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. >
What if I wanted to create different "incoming-sip" contexts depending on the service being used or number being called? For example sip calls coming from ipkall goto one context that presents a menu, but another sip call coming from one of the free German services provides a different menu and in German. Thanks, Rob _______________________________________________ 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
