So to clarify... Incoming calls to my SIP trunk, via Unlimitel, when the caller enters the extension (which is a person working on the internet a few cities away using a SIP Softphone like X-Lite), will always have the call data go through the Asterisk box until the call is completed. It will NEVER pass the call to the extension so that the originating call from Unlimitel talks directly to the user's Software at the extension (by passing the Asterisk box). Correct?
And if I want to allow this, my Trunk provider must allow "reinvite" (in which case Unlimitel does not allow). Are there any other possible ways to accomplish this that you know of? Are you aware of any Trunk providers that allow reinvite? Regards, Chuck -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon P. Ditner Sent: Wednesday, August 09, 2006 7:05 AM To: Chuck Mariotti Cc: [email protected] Subject: Re: [on-asterisk] Passing to Softphone Extension from Asterisk I don't believe Unlimitel supports reinvite's yet. If it does though, the 'reinvite' option needs to be set to 'yes' for both the trunk and the client in sip.conf If you look at the network traffic while on the asterisk box, you shouldn't see much of anything if you do a tcpdump something like this as root: # tcpdump -i <your external network interface, likely eth0> not port 5060 and host <ip address of client> On 8/9/06, Chuck Mariotti <[EMAIL PROTECTED]> wrote: > I have converted one of the unlimitel trunks to SIP and the trunk is > now operating. > > Do you have any idea of how to test if passing the call to a remote > SIP Softphone client is working? Is there a test or something I can do > to verify this is happening now? Or does it just "happen" automatically? > > Regards, > > Chuck > > -----Original Message----- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 08, 2006 2:05 PM > To: [email protected] > Subject: Re: [on-asterisk] Passing to Softphone Extension from > Asterisk > > On Tuesday 08 August 2006 13:23, Chuck Mariotti wrote: > > I am using unlimitel (IAX2 Trunk). > > ... > > > Extensions are configured as SIP extensions and we are currently > > using > > > X-Lite softphones. > > > internet, to her softphone). How would I make it so that the > > extension > > > talks straight to unlimitel and not through Asterisk? > > Not going to happen. You have IAX2 on one side and SIP on the other; > these two cannot talk to each other without something inbetween which > can translate. > > This is much the same as one leg of a call using ulaw and the other > leg using g729; the two will never be able to communicate directly > unless they have a common codec. Similarly, your two endpoints will > never communicate directly unless they have a common transport. > > Either use SIP with Unlimitel or get an Asterisk box at the remote > office. > > -A. > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] For additional > commands, e-mail: [EMAIL PROTECTED] > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] For additional > commands, e-mail: [EMAIL PROTECTED] > > -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca
