SIP URI calls

[TDF]

I notice when I make a call to a sip uri, nothing appears in the cdr, in the logs or in the freepbx interface, its as if the call never happens.

I don't know if this is desired or normal, is it a shortcoming in asterisk, is there a way around it ?

edit/ I was assuming a sip call from a softphone registered to PIAF isn't bypassing it, I think it might well be.

 

[blanchae]

You can make end call to end call by dialing directly to the destination URL, SIP allows that.

SIP is used to setup the call, RTP is used to transfer the data. If you know how to get there (URL) then you don't need Asterisk.

 

[TDF]

I'd prefer calls to go through asterisk.

I found a lot of unanswered questions on it from freepbx users, but did find a asterisk user with a solution

http://macnugget.org/projects/asterisk/page7

no idea how to translate that to freepbx.

 

[jmullinix]

Why not set up a custom extension in FreePBX and in the dial block put:

sip/[email protected]

Note that this is not a full SIP URI as it is missing the colon and one of the slashes. Change the pertinent information to work for you, otherwise, everytime you dial that extension, you will call me.

 

[TDF]

I did consider that, but really was thinking more of one off or irregularly dialled numbers.